Re: [cisco-voip] 2 different jabber TCT devices for one CUWL Standard User
It wouldn't do what I think you're trying to do. Is it possible to put both dn's on the same dual mode profile? Sent from my iPad On Feb 17, 2014, at 1:53 PM, Anthony Kouloglou ak...@dataways.gr wrote: Hi all, what would be the effect of creating 2 different TCT devices, assigning the same owner and associating this end user with both devices? Assume that ELM has a CUWL standard license available and the platforms are Cisco IM P 9.1 and CUCM 9.1. The idea is to have 2 different jabber numbers: one for an iphone for US one for an iphone for Europe. Cheers, Anthony ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat
Is it an option to use LDAP integration? There are certain cups features that won't work with out it. In a lab environment, you can always use OpenLDAP or something like that. Sent from my iPad On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Jason – I appreciate your response. As to your first question, yes, I have setup CCMCIP. From my original posting Application - Legacy Clients - CCMCIP Profile. I created a new profile, set the Primary and Secondary CCMCIP Host to be the IP Address of the CUCM Publisher. For Server Certification Verfication, I selected Any Certificate from the drop down. I assigned this profile to all 4 of my users. When I mouse over a contact, or right click and select View Profile, there is no indication of an IM address at all. If I select Edit Contact, I do not have the ability to enter an address into the Instant Messaging Address window. For all of my users, I have had to manually create the contacts as I am not using LDAP integration. This is why Im not able to initiate a chat – but the question is – why am I not able to manually enter the IM address like I can the DN? Im also not using an AD domain, so under CUPS - System - Cluster Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM enterprise parameters). Im beginning to wonder if CUPS is so dependent upon DNS and LDAP that not having them available is causing my failures. As for the log files, yes, I have looked into the files in CUPC8 - Local Settings and CUPC8 - App Data. Under Client Services Framework - Local Settings - Logs, I have searched for “error” and it only matches against entries for video. I have not attempted any Wireshark captures yet. Dr. Jeffrey T. Girard (Jeff), PhD Colonel, United States Army (Retired) Senior Network Engineer / VoIP Engineer - WireMeHappy.com reply to: jeffrey.gir...@wiremehappy.com (607)835-0406 (home office) (845)764-1661 (mobile) (607)835-0458 (fax) From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com] Sent: Saturday, March 15, 2014 7:40 PM To: Jeffrey Girard; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat In IMP do you have a CCMCIP setup? When you put a mouse over a contact or select the Contact Details what is the IM address? It should match your Presence Domain. Check the log files/Wireshark. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jeffrey Girard Sent: Friday, March 14, 2014 6:12 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat Home Lab – learning Presence CUCM 9 CUPS 9 (not 8.6 as virtually all of the help and discussion forums, documents, etc are for) CUPC 8.6 (3 copies running on 3 different laptops) 4 users configured in CUCM for Presence capabilities No LDAP integration Current status: Presence indicators for phones works fine, no issues 3 users can log into their CUPC clients and Presence information from their own associated deskphone displays correctly (ie if User 1 takes his deskphone offhook, the CUPC for User 1 shows as “On the Phone” Users can place phone calls with each other by typing the DN in the “Search for name or number” field I have two different issues that I have not been able to solve all day – they may be related to each other. The first issue: I am unable to go into deskphone mode. Although the checkbox is visible – it appears to be greyed out and clicking in it does nothing. The second issue: I am unable to start a chat session. I highlite a contact (that I manually entered), right click, and select chat. I get an error message that says “Failed to start conversation. Invalid parameter” I have scoured the Cisco site for docs and most of them pertain to CUPS 8.6 and not 9. There is a difference in that v9 does not have the Application - IP Phone Messenger selection. My current configs: CUCM 4 application users – CUPS-AXL (with Stand CCM SuperUsers permissions), CUPS-Deskphone, CUPS-CTIGW, and CUPS-PhoneMSG (all with Standard CTI Allow control of all devices). 3 end users – From top to bottom, all users have passwords and digest credentials. Under Service Settings, all users have the “Enable user for Unified CM IM and Presence” checked. This is the replacement for the assigning license capabilities from version 8.6. Also under Service Settings, all users have a UC Service Profile assigned. The UC Service Profile has two UC service settings – a Presence and IM Profile and a CTI Profile.
Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat
H ... CUPC is old tech . Let me fire up the not-so-way-back machine. Device and client control almost always comes down to an ownership issue 1. End User has Standard CTI Enabled privilege 2. User owns the Client Services Framework profile 3. User owns the actual phone (unless just a CSF user) 4. User is associated to the DN, of the prime line on the Phone/CSF 5. The Primary Phone field in the phone/CSF config is set to the Mac of the phone the user owns (only if there is an actual phone for the user) 6. End User has primary extension field set to the prime line of the CSF / Phone config 8. For kicks, play with the end user permissions too if none of the above works, give the user all permissions and see what that does. Let me know how it works out for you. Sent from my iPad On Mar 19, 2014, at 9:51 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Update – Well, I finally bit the bullet and stood up a MS AD domain in my lab. I integrated CUCM into the LDAP and imported the users from LDAP I also stood up a DNS server. I reconfigured CUPS to the new domain, joined all of the endpoint laptops to the domain, and retested. Same place as I was before with the deskphone mode – however, my ability to IM is fixed (as was expected). Using CUPS, I am not able to go into deskphone control mode. The option box is still greyed out. However, I installed Jabber for Windows on the same laptops. In Jabber, I am able to select the option at the bottom of the window to use my deskphone for calls, and then I can move it back to the Jabber client. So, anyone have any ideas why CUPC refuses to let me go into deskphone control mode? All features and functions of the JFW work great. No problems. Jeff From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Saturday, March 15, 2014 10:43 PM To: Jeffrey Girard Cc: Jason Aarons (AM); cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat Is it an option to use LDAP integration? There are certain cups features that won't work with out it. In a lab environment, you can always use OpenLDAP or something like that. Sent from my iPad On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Jason – I appreciate your response. As to your first question, yes, I have setup CCMCIP. From my original posting Application - Legacy Clients - CCMCIP Profile. I created a new profile, set the Primary and Secondary CCMCIP Host to be the IP Address of the CUCM Publisher. For Server Certification Verfication, I selected Any Certificate from the drop down. I assigned this profile to all 4 of my users. When I mouse over a contact, or right click and select View Profile, there is no indication of an IM address at all. If I select Edit Contact, I do not have the ability to enter an address into the Instant Messaging Address window. For all of my users, I have had to manually create the contacts as I am not using LDAP integration. This is why Im not able to initiate a chat – but the question is – why am I not able to manually enter the IM address like I can the DN? Im also not using an AD domain, so under CUPS - System - Cluster Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM enterprise parameters). Im beginning to wonder if CUPS is so dependent upon DNS and LDAP that not having them available is causing my failures. As for the log files, yes, I have looked into the files in CUPC8 - Local Settings and CUPC8 - App Data. Under Client Services Framework - Local Settings - Logs, I have searched for “error” and it only matches against entries for video. I have not attempted any Wireshark captures yet. Dr. Jeffrey T. Girard (Jeff), PhD Colonel, United States Army (Retired) Senior Network Engineer / VoIP Engineer - WireMeHappy.com reply to: jeffrey.gir...@wiremehappy.com (607)835-0406 (home office) (845)764-1661 (mobile) (607)835-0458 (fax) From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com] Sent: Saturday, March 15, 2014 7:40 PM To: Jeffrey Girard; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat In IMP do you have a CCMCIP setup? When you put a mouse over a contact or select the Contact Details what is the IM address? It should match your Presence Domain. Check the log files/Wireshark. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jeffrey Girard Sent: Friday, March 14, 2014
Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat
Well at this point is say get a TAC case opened if it's for production Sent from my iPhone On Mar 20, 2014, at 9:14 AM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Replies inline…. Jeff From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Wednesday, March 19, 2014 10:21 PM To: Jeffrey Girard Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat H ... CUPC is old tech . Let me fire up the not-so-way-back machine. Device and client control almost always comes down to an ownership issue 1. End User has Standard CTI Enabled privilege Yes 2. User owns the Client Services Framework profile Yes 3. User owns the actual phone (unless just a CSF user) Yes 4. User is associated to the DN, of the prime line on the Phone/CSF Yes 5. The Primary Phone field in the phone/CSF config is set to the Mac of the phone the user owns (only if there is an actual phone for the user) Yes 6. End User has primary extension field set to the prime line of the CSF / Phone config Yes. Configured in Active Directory, imported via LDAP, confirmed on End User Page 7. For kicks, play with the end user permissions too if none of the above works, give the user all permissions and see what that does. Assigned the user to all privilege groups. Went to CUPS and restarted the DirSync service. Closed out of the CUPS application and restarted. No change. In CUPC, I am unable to go into deskphone control mode – the option is displayed at the bottom of the window, but its greyed out. As with yesterday, JFW work great. So, I am baffled. Let me know how it works out for you. Sent from my iPad On Mar 19, 2014, at 9:51 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Update – Well, I finally bit the bullet and stood up a MS AD domain in my lab. I integrated CUCM into the LDAP and imported the users from LDAP I also stood up a DNS server. I reconfigured CUPS to the new domain, joined all of the endpoint laptops to the domain, and retested. Same place as I was before with the deskphone mode – however, my ability to IM is fixed (as was expected). Using CUPS, I am not able to go into deskphone control mode. The option box is still greyed out. However, I installed Jabber for Windows on the same laptops. In Jabber, I am able to select the option at the bottom of the window to use my deskphone for calls, and then I can move it back to the Jabber client. So, anyone have any ideas why CUPC refuses to let me go into deskphone control mode? All features and functions of the JFW work great. No problems. Jeff From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Saturday, March 15, 2014 10:43 PM To: Jeffrey Girard Cc: Jason Aarons (AM); cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat Is it an option to use LDAP integration? There are certain cups features that won't work with out it. In a lab environment, you can always use OpenLDAP or something like that. Sent from my iPad On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Jason – I appreciate your response. As to your first question, yes, I have setup CCMCIP. From my original posting Application - Legacy Clients - CCMCIP Profile. I created a new profile, set the Primary and Secondary CCMCIP Host to be the IP Address of the CUCM Publisher. For Server Certification Verfication, I selected Any Certificate from the drop down. I assigned this profile to all 4 of my users. When I mouse over a contact, or right click and select View Profile, there is no indication of an IM address at all. If I select Edit Contact, I do not have the ability to enter an address into the Instant Messaging Address window. For all of my users, I have had to manually create the contacts as I am not using LDAP integration. This is why Im not able to initiate a chat – but the question is – why am I not able to manually enter the IM address like I can the DN? Im also not using an AD domain, so under CUPS - System - Cluster Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM enterprise parameters). Im beginning to wonder if CUPS is so dependent upon DNS and LDAP that not having them available is causing my failures. As for the log files, yes, I have looked into the files in CUPC8 - Local Settings and CUPC8 - App Data. Under Client Services Framework - Local Settings - Logs, I have searched for “error” and it only matches against entries for video
Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat
At this point if say get a TAC case opened if it is production and you have a contract/support. I retired my use of CUPC and moved to JFW . CUPC should be working as long as all the ownership stuff is setup correctly. Sent from my iPhone On Mar 20, 2014, at 9:14 AM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Replies inline…. Jeff From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Wednesday, March 19, 2014 10:21 PM To: Jeffrey Girard Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat H ... CUPC is old tech . Let me fire up the not-so-way-back machine. Device and client control almost always comes down to an ownership issue 1. End User has Standard CTI Enabled privilege Yes 2. User owns the Client Services Framework profile Yes 3. User owns the actual phone (unless just a CSF user) Yes 4. User is associated to the DN, of the prime line on the Phone/CSF Yes 5. The Primary Phone field in the phone/CSF config is set to the Mac of the phone the user owns (only if there is an actual phone for the user) Yes 6. End User has primary extension field set to the prime line of the CSF / Phone config Yes. Configured in Active Directory, imported via LDAP, confirmed on End User Page 7. For kicks, play with the end user permissions too if none of the above works, give the user all permissions and see what that does. Assigned the user to all privilege groups. Went to CUPS and restarted the DirSync service. Closed out of the CUPS application and restarted. No change. In CUPC, I am unable to go into deskphone control mode – the option is displayed at the bottom of the window, but its greyed out. As with yesterday, JFW work great. So, I am baffled. Let me know how it works out for you. Sent from my iPad On Mar 19, 2014, at 9:51 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Update – Well, I finally bit the bullet and stood up a MS AD domain in my lab. I integrated CUCM into the LDAP and imported the users from LDAP I also stood up a DNS server. I reconfigured CUPS to the new domain, joined all of the endpoint laptops to the domain, and retested. Same place as I was before with the deskphone mode – however, my ability to IM is fixed (as was expected). Using CUPS, I am not able to go into deskphone control mode. The option box is still greyed out. However, I installed Jabber for Windows on the same laptops. In Jabber, I am able to select the option at the bottom of the window to use my deskphone for calls, and then I can move it back to the Jabber client. So, anyone have any ideas why CUPC refuses to let me go into deskphone control mode? All features and functions of the JFW work great. No problems. Jeff From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Saturday, March 15, 2014 10:43 PM To: Jeffrey Girard Cc: Jason Aarons (AM); cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat Is it an option to use LDAP integration? There are certain cups features that won't work with out it. In a lab environment, you can always use OpenLDAP or something like that. Sent from my iPad On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com wrote: Jason – I appreciate your response. As to your first question, yes, I have setup CCMCIP. From my original posting Application - Legacy Clients - CCMCIP Profile. I created a new profile, set the Primary and Secondary CCMCIP Host to be the IP Address of the CUCM Publisher. For Server Certification Verfication, I selected Any Certificate from the drop down. I assigned this profile to all 4 of my users. When I mouse over a contact, or right click and select View Profile, there is no indication of an IM address at all. If I select Edit Contact, I do not have the ability to enter an address into the Instant Messaging Address window. For all of my users, I have had to manually create the contacts as I am not using LDAP integration. This is why Im not able to initiate a chat – but the question is – why am I not able to manually enter the IM address like I can the DN? Im also not using an AD domain, so under CUPS - System - Cluster Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM enterprise parameters). Im beginning to wonder if CUPS is so dependent upon DNS and LDAP that not having them available is causing my failures. As for the log files, yes, I have looked into the files in CUPC8 - Local Settings and CUPC8 - App
Re: [cisco-voip] CUC 8.6.2.20000 Import from CUCM takes 15 minutes...
Unified reporting... - Does the cluster and db replication show healthy? Unified serviceability - bounce the BAT service Sent from my iPhone On Mar 31, 2014, at 1:29 AM, Jonathan Charles jonv...@gmail.com wrote: Repeatable; just started... can add users locally, but cannot import from CUCM... searching takes forever, and once you find them, the import takes forever as well (realistically, about 15 minutes for each...) Any ideas? Jonathan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUWL 10.x licenses
If you're talking about CUCM and CUC; in the 9.x branch Cisco introduced the ELM (Enterprise License Manager). When you move into the 10.x's you have to use the PLM (Prime License Manager). If your licensing is on the ELM and you want to goto the PLM (for 10.x) then you have to get fancy with TAC. They can deduct the amount of licensing that you need from the ELM and re-host it for the PLM (but you have to provide them with the license request from the PLM). So lets say you have 200 CUWL Pro licenses on an ELM for UCOS 9.1.x and you want 100 of them to goto a PLM for UCOS 10.x. TAC can take your license from the ELM and re-host it minus 100 (leaving you with 100 CUWL Pro in the ELM), then TAC can take the extra 100 and re-host them into the PLM as CUWL Pro for UCOS 10.X. If you used the PLM with 9.1.X and have upgraded UCOS to 10.X from 9.1.X then the licenses need to be re-hosted. Prove your upgrade entitlement to TAC and ask them to re-host the licenses. Ryan Huff CCNA R/S, CCNA Wireless, CCNA Voice, CCNP Voice, UC on UCSOS, UCCX Specialist, CCIE Collaboration (written) From: peders...@bennettjones.com To: mloradi...@heliontechnologies.com; cisco-voip@puck.nether.net Date: Mon, 22 Sep 2014 21:12:34 + Subject: Re: [cisco-voip] CUWL 10.x licenses Interesting that Cisco switched to a manual process. I opened an SR and the guy from Licensing said I should contact The Product Manager whatever that means. Good thing I get 60 days to sort this out J From: Matthew Loraditch [mailto:mloradi...@heliontechnologies.com] Sent: 22 September 2014 1:56 PM To: Eric Pedersen; cisco-voip (cisco-voip@puck.nether.net) Subject: RE: CUWL 10.x licenses You submit a manual case for all major upgrades now. Just like for 8 to 9. Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA 1965 Greenspring Drive Timonium, MD 21093 direct voice. 443.541.1518 fax. 410.252.9284 Twitter | Facebook | Website | Email Support Support Phone. 410.252.8830 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Eric Pedersen Sent: Monday, September 22, 2014 3:48 PM To: cisco-voip (cisco-voip@puck.nether.net) Subject: [cisco-voip] CUWL 10.x licenses We upgraded to CUCX 10.5 from 9.1 on the weekend and the 9.x licenses installed on PLM need to be upgraded. Does anyone know how to get a PAK for this? I didn't receive any licenses in the upgrade e-delivery for CUCM and CUCX. Thanks, Eric The contents of this message may contain confidential and/or privileged subject matter. If this message has been received in error, please contact the sender and delete all copies. Like other forms of communication, e-mail communications may be vulnerable to interception by unauthorized parties. If you do not wish us to communicate with you by e-mail, please notify us at your earliest convenience. In the absence of such notification, your consent is assumed. Should you choose to allow us to communicate by e-mail, we will not take any additional security measures (such as encryption) unless specifically requested. If you no longer wish to receive commercial messages, you can unsubscribe by accessing this link: http://www.bennettjones.com/unsubscribe The contents of this message may contain confidential and/or privileged subject matter. If this message has been received in error, please contact the sender and delete all copies. Like other forms of communication, e-mail communications may be vulnerable to interception by unauthorized parties. If you do not wish us to communicate with you by e-mail, please notify us at your earliest convenience. In the absence of such notification, your consent is assumed. Should you choose to allow us to communicate by e-mail, we will not take any additional security measures (such as encryption) unless specifically requested. If you no longer wish to receive commercial messages, you can unsubscribe by accessing this link: http://www.bennettjones.com/unsubscribe ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Delete Log Files
Martin, If you have a space concern (trying to do an in-place upgrade etc), you can adjust the High/Low logging watermarks but remember to set them back after you're finished! Thanks, Ryan Huff CCNA R/S, CCNA Wireless, CCNA Voice, CCNP Voice, UC on UCS Specialist, UCCX Product Specialist, CCIE Collaboration (Written) Date: Tue, 23 Sep 2014 18:19:38 +0530 From: sknt...@gmail.com To: m...@bilobit.com CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Delete Log Files Martin, Is there a reason you'd like to do this? What is the use case?By default, the older traces will get overwritten when there is no space left for newer traces.You can manually delete the files from the CLI or the Remote Browse on RTMT. ThanksSreekanth On 23 September 2014 15:43, Martin Schmuker m...@bilobit.com wrote: Guys, is there any way to delete CUCM log files (aka traces) after x days? Thanks, Martin ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] MSP VoIP Monitoring
Ryan, Prime is pretty sweet but if you're looking for qos tools (jitter stats, mos scores, delays ... etc) then I would look at two products: 1.) LiveAction . very qos centric 2.) NetBrains .. does do a lot with QOS but also does general layer2/3/4 mapping We're eval'ing NetBrains now and it looks pretty sweet. For instance, I can open a mapped Visio drawing of my topo that NetBrains created and I can hover on an interface and it will show any jitter/delays outside the threshold I define From: dennis.h...@wwt.com To: rburt...@gmail.com Date: Thu, 25 Sep 2014 12:04:22 -0500 CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] MSP VoIP Monitoring Prime Collab with Assurance and Analytics would come to mind. I don’t have first-hand experience with that piece to the Prime Pie. Dennis Heim | Collaboration Solutions ArchitectWorld Wide Technology, Inc. | +1 314-212-1814 From: Ryan Burtch [mailto:rburt...@gmail.com] Sent: Thursday, September 25, 2014 1:00 PM To: Heim, Dennis Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] MSP VoIP Monitoring Dennis: We have VOSS and we use that to admin our HCS system. The thing we need is voice traffic monitoring for our on-prem customers in one system. This needs to give us the ability to see what is going on with the VoIP specific traffic. e.g. who has excessive jitter on their line, dropped calls, MOS scores, etc. Any Ideas? Sincerely, Ryan Burtch On Thu, Sep 25, 2014 at 12:53 PM, Heim, Dennis dennis.h...@wwt.com wrote:There is also Voss-4-UC, which has RBAC and multi-tenancy. Dennis Heim | Collaboration Solutions ArchitectWorld Wide Technology, Inc. | +1 314-212-1814 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Burtch Sent: Thursday, September 25, 2014 11:09 AM To: cisco-voip@puck.nether.net Subject: [cisco-voip] MSP VoIP Monitoring My company is looking for a VoIP monitoring solution that is Multi-Tennant capable. We manage several Customer's Voice environment and we need to be able to provide them statistics, troubleshooting, etc through a single pane of glass. This would be something like multi-tennant SolarWinds. Does anyone have any ideas? Sincerely, Ryan Burtch ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Phone Configuration
The only truly supported call manager way of barging a connected call is barge or cbarge, but they both require a 'button push'. Now if your connected call is offnet, you could potentially do a PLAR on the client phone, so as soon as the off-hook event triggers on the client phone it dials into the offnet conference. If the connected call in onnet, then barge or cbarge are really your options. If you want to venture into layer 2 of the OSI model however, you could do a SPAN on the ports of the phones that are doing the connected calls and then dump the RTP/signaling stream to the port of the client phone. Not sure if that would work or not, you'd just have to play with it a bit. Thanks, Ryan From: nh...@co.fresno.ca.us To: cisco-voip@puck.nether.net Date: Tue, 30 Sep 2014 22:41:57 + Subject: [cisco-voip] Phone Configuration We use several Translations services, is there a way to configure a phone to join into a phone call without pushing anything? Basically, our Employees call translation service and gives the account information, then gestures to the client to pick up the phone, at this point the client phone just joins into the conference without the customer pushing any buttons. Is this possible? Neal Haas IT Analyst, Communications ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits
So on router B, do the following and then place an inbound call from a known number: hostname# conf t hostname (config)# logging console hostname (config)# exit hostname# term mon hostname# debug isdn q931 What do you see for Calling Party Number i =, Plan: and Type:? The plan and type fields should be located below the calling party number field. Once you've determined that you really are receiving 4 digits from the telco on that circuit the next step is to figure out what in the router config is stripping the digit. If you find that you are not receiving 4 digits from the telco, you can either work with the telco to fix it on the PRI or if the missing digit is a constant, you can add it with an inbound voice translation rule. Look at your inbound dial peers; do you have any voice translation rules that are doing any digit stripping? Are the inbound voice translations using different regex than on the router that is working and if so, what is different? Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice CCNA Route/Switch, CCNA Wireless, UCCX Specialist From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Sun, 16 Nov 2014 08:18:22 + Subject: [cisco-voip] E1 CAS card in Mexico not getting enough digits I have a problem with a E1 CAS site in Mexico. I have two routers. Router B is problem. In router A I see 4 digits come in on DTMF. In router B I see 3 digits come in on DTMF. We are missing 1 digit. I swap the circuit and works fine in router A. Same IOS both routers. Controller e1 0/0/0 Framing NO-CRC4 dso-group 0 timeslots 1-10 type r2-digital dtmf dnis cas-custom 0 Country telmex Seizure-ackt-time 2 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits
Ahh, missed the part about CAS, sorry. So is the provider doing something on their side with the MAC of the E1 on your side? If the provider is sending 3 digits (regardless if it works on the other router) then it should be their issue. You have the plan and type set according to what the provider expects? Are you getting 3 digits for the calling or called party? From: jason.aar...@dimensiondata.com To: ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits Date: Sun, 16 Nov 2014 19:56:50 + CAS no q931 Debug shows on bad router we receive 3 digits from provider, move circuit to another router we receive 4 digits from provider. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Sunday, November 16, 2014 8:52 AM To: Jason Aarons (AM); cisco-voip@puck.nether.net Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits So on router B, do the following and then place an inbound call from a known number: hostname# conf t hostname (config)# logging console hostname (config)# exit hostname# term mon hostname# debug isdn q931 What do you see for Calling Party Number i =, Plan: and Type:? The plan and type fields should be located below the calling party number field. Once you've determined that you really are receiving 4 digits from the telco on that circuit the next step is to figure out what in the router config is stripping the digit. If you find that you are not receiving 4 digits from the telco, you can either work with the telco to fix it on the PRI or if the missing digit is a constant, you can add it with an inbound voice translation rule. Look at your inbound dial peers; do you have any voice translation rules that are doing any digit stripping? Are the inbound voice translations using different regex than on the router that is working and if so, what is different? Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice CCNA Route/Switch, CCNA Wireless, UCCX Specialist From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Sun, 16 Nov 2014 08:18:22 + Subject: [cisco-voip] E1 CAS card in Mexico not getting enough digits I have a problem with a E1 CAS site in Mexico. I have two routers. Router B is problem. In router A I see 4 digits come in on DTMF. In router B I see 3 digits come in on DTMF. We are missing 1 digit. I swap the circuit and works fine in router A. Same IOS both routers. Controller e1 0/0/0 Framing NO-CRC4 dso-group 0 timeslots 1-10 type r2-digital dtmf dnis cas-custom 0 Country telmex Seizure-ackt-time 2 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip itevomcid ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits
Well it had to either be the Telco or an inbound xlate. Glad you found it! Good job, now go have an Iced Tea and a vacation! From: jason.aar...@dimensiondata.com To: avholloway+cisco-v...@gmail.com; ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits Date: Mon, 17 Nov 2014 06:03:23 + So after about 5 CCIEs looked at it and 5 Cisco AS onsite engineers and 2 days of effort we found the translation-profile on the voice-port was somehow responsible. Seems the incoming digits took longer on router-b. Go figure. We move the translation-profile from the voice port to the dial-peer. I hate CAS. From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] Sent: Sunday, November 16, 2014 9:38 PM To: Jason Aarons (AM); Ryan Huff; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits What if...just what if, it's alternating between 3 and 4 digits every time you unplug it? Try unplugging it and re-plugging it into the same router to validate this crazy idea. On Sun Nov 16 2014 at 7:21:11 PM Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: If you move circuit to another router and it works, then its hard to blame the carrier! CAS stinks. Sent from my Verizon Wireless 4G LTE Smartphone Original message From: Ryan Huff Date:11/16/2014 15:45 (GMT-05:00) To: Jason Aarons (AM) , cisco-voip@puck.nether.net Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits Ahh, missed the part about CAS, sorry. So is the provider doing something on their side with the MAC of the E1 on your side? If the provider is sending 3 digits (regardless if it works on the other router) then it should be their issue. You have the plan and type set according to what the provider expects? Are you getting 3 digits for the calling or called party? From: jason.aar...@dimensiondata.com To: ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits Date: Sun, 16 Nov 2014 19:56:50 + CAS no q931 Debug shows on bad router we receive 3 digits from provider, move circuit to another router we receive 4 digits from provider. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Sunday, November 16, 2014 8:52 AM To: Jason Aarons (AM); cisco-voip@puck.nether.net Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits So on router B, do the following and then place an inbound call from a known number: hostname# conf t hostname (config)# logging console hostname (config)# exit hostname# term mon hostname# debug isdn q931 What do you see for Calling Party Number i =, Plan: and Type:? The plan and type fields should be located below the calling party number field. Once you've determined that you really are receiving 4 digits from the telco on that circuit the next step is to figure out what in the router config is stripping the digit. If you find that you are not receiving 4 digits from the telco, you can either work with the telco to fix it on the PRI or if the missing digit is a constant, you can add it with an inbound voice translation rule. Look at your inbound dial peers; do you have any voice translation rules that are doing any digit stripping? Are the inbound voice translations using different regex than on the router that is working and if so, what is different? Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice CCNA Route/Switch, CCNA Wireless, UCCX Specialist From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Sun, 16 Nov 2014 08:18:22 + Subject: [cisco-voip] E1 CAS card in Mexico not getting enough digits I have a problem with a E1 CAS site in Mexico. I have two routers. Router B is problem. In router A I see 4 digits come in on DTMF. In router B I see 3 digits come in on DTMF. We are missing 1 digit. I swap the circuit and works fine in router A. Same IOS both routers. Controller e1 0/0/0 Framing NO-CRC4 dso-group 0 timeslots 1-10 type r2-digital dtmf dnis cas-custom 0 Country telmex Seizure-ackt-time 2 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip itevomcid ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] HELP
Well, you can't make calls because there are no resources available to negotiate the call, that much is clear. I assume this issue came about after you upgraded the IOS on the router? What platform do you have and what code are you currently running? The firmware for the DSP is in the IOS code itself. As long as you're on supported code for the platform, activate the DSP Farm and then reboot the router. This should resolve the issue. Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice CCNA R/S, CCNA Wireless UCCX Specialist Date: Mon, 17 Nov 2014 13:42:03 +0100 From: eteng.o...@unicem.com.ng To: cisco-voip@puck.nether.net Subject: [cisco-voip] HELP HELP,I CAN'T CALL OUT ON E1 AND FXO. SEE MESSAGE ON THE ROUTER. voice-card 0 ! Warning! DSPs 5 in slot 0 are using non-default firmware from device flash: ! This is not recommended, the IOS default version is 24.3.3 If you are not the intended recipient and have received this e-mail in error, please delete it immediately from your system and notify the sender by e-mail or telephone. Be notified that you are not to copy, distribute or disclose this e-mail or any of its contents to any other party and any such action may be unlawful. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] VMware 5.5
Martin, Have you disabled Large Receive Offset (LRO) on all the Elastic Sky hosts? http://docwiki.cisco.com/wiki/Disable_LRO Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice, CCNA Route Switch, CCNA Wireless UCCX Specialist From: m...@bilobit.com To: james.buchan...@gmail.com Date: Mon, 17 Nov 2014 18:06:51 + CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] VMware 5.5 So 5.5 is supported with CUCM 9.1.x So the question is: Why do I have CUCM at 100% CPU? Only hard reset helps at this point. Thanks, Martin From: James Buchanan [mailto:james.buchan...@gmail.com] Sent: Monday, November 17, 2014 1:10 PM To: Martin Schmuker Cc: Cisco VoIP Mailing List Subject: Re: [cisco-voip] VMware 5.5 Hello, Whenever you are looking for answers on UC virtualization, look here: http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29 Thanks, James On Mon, Nov 17, 2014 at 5:36 AM, Martin Schmuker m...@bilobit.com wrote: Guys, since 3 weeks we are running our UC Environment (CUCM, Unity Cxn, IMP) on ESXi 5.5. At Friday we setup new vCenter 5.5 and added the esx hosts. Since saturday, all machines are stuck in 100% CPU after a few hours. Sometimes all machines at the *same* time! They don’t reply Ping (ICMP Echo), and CPU is at 100%. CUCM and Cxn are on 9.1(2)SU2a (9.1.2.12901-3). Someone has any idea? Is 9.1.x not supported on vSphere 5.5? Thanks, Martin ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] VMware 5.5
While that particular DocWiki does suggest that it is no longer needed to disable Large Receive Offset on ESXi 4.1 with CUCUM 8.6 and above; you should also note that one of the specific issues having LRO on can cause is consistent CPU pegging for the UCOS guests (wouldn't be the first time something didn't work the way Cisco's docs say it should). Please also review: https://supportforums.cisco.com/document/95886/disable-lro-ucs-uc-application-deployments Again, I can't say beyond a shadow of doubt that this is the issue anymore than anyone can say that it isn't. I would also look into the RAID drivers as well as all the physical connections on the UCS boxes. Thanks, rh From: wo...@justfamily.org Date: Mon, 17 Nov 2014 14:30:21 -0700 Subject: Re: [cisco-voip] VMware 5.5 To: ryanh...@outlook.com CC: m...@bilobit.com; james.buchan...@gmail.com; cisco-voip@puck.nether.net That page specifically says you don't have to disable LRO if you are above 4.1 esxi with8.6 CUCM. Martin, which version of 5.5 are you on? GA, update 1 or 2? I recently ran into an issue with a customer on 5.5 u1 and too new of raid drivers on the card and had to downgrade the drivers. Wasn't causing 100% cpu, but very slow drive access times. So checked your vmware version and compatibility on the hardware you are running. On Mon, Nov 17, 2014 at 11:59 AM, Ryan Huff ryanh...@outlook.com wrote: Martin, Have you disabled Large Receive Offset (LRO) on all the Elastic Sky hosts? http://docwiki.cisco.com/wiki/Disable_LRO Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice, CCNA Route Switch, CCNA Wireless UCCX Specialist From: m...@bilobit.com To: james.buchan...@gmail.com Date: Mon, 17 Nov 2014 18:06:51 + CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] VMware 5.5 So 5.5 is supported with CUCM 9.1.x So the question is: Why do I have CUCM at 100% CPU? Only hard reset helps at this point. Thanks, Martin From: James Buchanan [mailto:james.buchan...@gmail.com] Sent: Monday, November 17, 2014 1:10 PM To: Martin Schmuker Cc: Cisco VoIP Mailing List Subject: Re: [cisco-voip] VMware 5.5 Hello, Whenever you are looking for answers on UC virtualization, look here: http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29 Thanks, James On Mon, Nov 17, 2014 at 5:36 AM, Martin Schmuker m...@bilobit.com wrote: Guys, since 3 weeks we are running our UC Environment (CUCM, Unity Cxn, IMP) on ESXi 5.5. At Friday we setup new vCenter 5.5 and added the esx hosts. Since saturday, all machines are stuck in 100% CPU after a few hours. Sometimes all machines at the *same* time! They don’t reply Ping (ICMP Echo), and CPU is at 100%. CUCM and Cxn are on 9.1(2)SU2a (9.1.2.12901-3). Someone has any idea? Is 9.1.x not supported on vSphere 5.5? Thanks, Martin ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] VMware 5.5
I had a CPU pegging issue about 4.5 months ago on ESXi 5.0. I turned LRO off (only change I made), and I haven't had an issue with CPU pegging since. My issue was due to the TCP throughput being irregular on VMXNET2 and VMXNET3 and it was causing HELLO/ACK issues with the database replication (it would get into a setup loop and never stop). Guest reboots didn't fix it, if a rebooted the UCS box it worked for a little bit and then went right back to the issue VMware states that the issue is known in Linux kernel 2.6.24 and later with VMXNET3 (http://kb.vmware.com/selfservice/microsites/search.do?language=en_UScmd=displayKCexternalId=1027511) There are plenty of reports on the Internet were the LRO/CUC issue has been found beyond esxi 4.1 and cucm 8.6. RHEL probably isn't going to get any performance gains from LRO anyhow so if it were me, I'd turn it off. It's broke now, right? Generally, you can't break broke so I would try it, if for no other reason than to say that isn't it. Again, I am not saying this IS your issue, just that it COULD BE a contributing factor, it would be worth a shot in my book if you haven't found the issue yet. From: wo...@justfamily.org Date: Mon, 17 Nov 2014 23:50:52 -0700 Subject: Re: [cisco-voip] VMware 5.5 To: ryanh...@outlook.com CC: m...@bilobit.com; james.buchan...@gmail.com; cisco-voip@puck.nether.net This discussion comes up about every 6 months on this list, LRO is no longer affected and was only a 4.1 issue on esxi. http://puck.nether.net/pipermail/cisco-voip/2012-October/029907.html http://docwiki.cisco.com/wiki/Unified_Communications_VMware_Requirements#Supported_Versions.2C_Patches_and_Updates_of_VMware_vSphere_ESXi The latter being definitive for me, they keep that doc updated pretty well and note, it states 4.1, not any other version of esxi. Martin has another issue. :) On Mon, Nov 17, 2014 at 4:46 PM, Ryan Huff ryanh...@outlook.com wrote: While that particular DocWiki does suggest that it is no longer needed to disable Large Receive Offset on ESXi 4.1 with CUCUM 8.6 and above; you should also note that one of the specific issues having LRO on can cause is consistent CPU pegging for the UCOS guests (wouldn't be the first time something didn't work the way Cisco's docs say it should). Please also review: https://supportforums.cisco.com/document/95886/disable-lro-ucs-uc-application-deployments Again, I can't say beyond a shadow of doubt that this is the issue anymore than anyone can say that it isn't. I would also look into the RAID drivers as well as all the physical connections on the UCS boxes. Thanks, rh From: wo...@justfamily.org Date: Mon, 17 Nov 2014 14:30:21 -0700 Subject: Re: [cisco-voip] VMware 5.5 To: ryanh...@outlook.com CC: m...@bilobit.com; james.buchan...@gmail.com; cisco-voip@puck.nether.net That page specifically says you don't have to disable LRO if you are above 4.1 esxi with8.6 CUCM. Martin, which version of 5.5 are you on? GA, update 1 or 2? I recently ran into an issue with a customer on 5.5 u1 and too new of raid drivers on the card and had to downgrade the drivers. Wasn't causing 100% cpu, but very slow drive access times. So checked your vmware version and compatibility on the hardware you are running. On Mon, Nov 17, 2014 at 11:59 AM, Ryan Huff ryanh...@outlook.com wrote: Martin, Have you disabled Large Receive Offset (LRO) on all the Elastic Sky hosts? http://docwiki.cisco.com/wiki/Disable_LRO Thanks, Ryan Huff CCIE Collaboration (Written), CCNP Voice, CCNA Voice, CCNA Route Switch, CCNA Wireless UCCX Specialist From: m...@bilobit.com To: james.buchan...@gmail.com Date: Mon, 17 Nov 2014 18:06:51 + CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] VMware 5.5 So 5.5 is supported with CUCM 9.1.x So the question is: Why do I have CUCM at 100% CPU? Only hard reset helps at this point. Thanks, Martin From: James Buchanan [mailto:james.buchan...@gmail.com] Sent: Monday, November 17, 2014 1:10 PM To: Martin Schmuker Cc: Cisco VoIP Mailing List Subject: Re: [cisco-voip] VMware 5.5 Hello, Whenever you are looking for answers on UC virtualization, look here: http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29 Thanks, James On Mon, Nov 17, 2014 at 5:36 AM, Martin Schmuker m...@bilobit.com wrote: Guys, since 3 weeks we are running our UC Environment (CUCM, Unity Cxn, IMP) on ESXi 5.5. At Friday we setup new vCenter 5.5 and added the esx hosts. Since saturday, all machines are stuck in 100% CPU after a few hours. Sometimes all machines at the *same* time! They don’t reply Ping (ICMP Echo), and CPU is at 100%. CUCM and Cxn are on 9.1(2)SU2a (9.1.2.12901-3). Someone has any idea? Is 9.1.x not supported on vSphere 5.5? Thanks, Martin ___ cisco-voip mailing list cisco-voip@puck.nether.net https
Re: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?
What version of CUCM? From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Fri, 21 Nov 2014 17:46:14 + Subject: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones? I had a coworker tell me he added a subscriber during the day and all phones reset during the day. Perhaps this was ITL update? He wasn't expecting all phone in cluster to reset until he added new sub to a callmanger group after hours. Any one else seen this? Is it expected? Sent from my Verizon Wireless 4G LTE Smartphone ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?
Never mind just saw the subject line! So when the phones reset, did any of them register to the new subscriber or was it just a phone reset? If it isn't in the CM group then no phones should have registered. Is this 10.5(1) or SU1 or SU2? An ITL reload sounds plausible. From: ryanh...@outlook.com To: jason.aar...@dimensiondata.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones? Date: Fri, 21 Nov 2014 13:00:40 -0500 What version of CUCM? From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Fri, 21 Nov 2014 17:46:14 + Subject: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones? I had a coworker tell me he added a subscriber during the day and all phones reset during the day. Perhaps this was ITL update? He wasn't expecting all phone in cluster to reset until he added new sub to a callmanger group after hours. Any one else seen this? Is it expected? Sent from my Verizon Wireless 4G LTE Smartphone ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] AD Sync
What version of CUCM and What version of AD? Has the AD sync ever worked correctly? Is this a break/fix?What changed from when it was working?Is the distinguished user the same, did that user's AD permissions changeDoes the distinguished user have the delegate control privilege on the domain?Is this a new install?Were any changes made to AD after the original full sync the first time?Has either the domain name of the CUCM cluster or the AD server changed since the first time the LDAP full sync was ran?Does the BIND authentication work correctly? Have you completely removed an existing user account and then re-synced from AD to see if that account re-appears? Date: Wed, 26 Nov 2014 16:09:51 +0530 From: sknt...@gmail.com To: shabbar_babraw...@hotmail.com CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] AD Sync Hi Shabbar, What is the CUCM version? So the users go into Inactive mode every 6 hours? Or once everyday? If once, what time does that happen and is that during a sync? Have you taken a look at the DirSync logs during the period of failure? What about a packet capture to see if this could be an issue due to the network? Thanks Sreekanth On 26 November 2014 at 11:37, shabbar babrawala shabbar_babraw...@hotmail.com wrote: Hi Have a strange problem where the sync with AD has broken , everyday morning we have to keep performing a full sync as the users show inactive even though the setting is to sync every 6 hours. Have even deleted the LDAP configuration and redone but no luck. any help is appreciated. Shabbar ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] AD Sync
Shabbar, Sounds like the LDAP Manger Distinguished User or the LDAP search base is acting strangely. Have you made any OU/permisison changes in AD? Try using a new/different distinguished manager user in the LDAP directory configuration. Also, are you using any custom filters? Thanks, Ryan From: shabbar_babraw...@hotmail.com To: ryanh...@outlook.com; sknt...@gmail.com CC: cisco-voip@puck.nether.net Subject: RE: [cisco-voip] AD Sync Date: Wed, 26 Nov 2014 14:42:30 + Hi Cucm 9.1 win 2012 It was working before on win 2003 broken after upgrade to 2012 Regards Shabbar From: ryanh...@outlook.com To: sknt...@gmail.com; shabbar_babraw...@hotmail.com CC: cisco-voip@puck.nether.net Subject: RE: [cisco-voip] AD Sync Date: Wed, 26 Nov 2014 09:38:59 -0500 What version of CUCM and What version of AD? Has the AD sync ever worked correctly? Is this a break/fix?What changed from when it was working?Is the distinguished user the same, did that user's AD permissions changeDoes the distinguished user have the delegate control privilege on the domain?Is this a new install?Were any changes made to AD after the original full sync the first time?Has either the domain name of the CUCM cluster or the AD server changed since the first time the LDAP full sync was ran?Does the BIND authentication work correctly? Have you completely removed an existing user account and then re-synced from AD to see if that account re-appears? Date: Wed, 26 Nov 2014 16:09:51 +0530 From: sknt...@gmail.com To: shabbar_babraw...@hotmail.com CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] AD Sync Hi Shabbar, What is the CUCM version? So the users go into Inactive mode every 6 hours? Or once everyday? If once, what time does that happen and is that during a sync? Have you taken a look at the DirSync logs during the period of failure? What about a packet capture to see if this could be an issue due to the network? Thanks Sreekanth On 26 November 2014 at 11:37, shabbar babrawala shabbar_babraw...@hotmail.com wrote: Hi Have a strange problem where the sync with AD has broken , everyday morning we have to keep performing a full sync as the users show inactive even though the setting is to sync every 6 hours. Have even deleted the LDAP configuration and redone but no luck. any help is appreciated. Shabbar ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] IMP external database front-end
Hi Charles, I got a super simple front end written in PHP running on the same LAMP stack the the PostGreSQL database is running on (benefit of being a former programmer). It isn't 100% complete but will give you a real-time flow of persistent chat messages and p 2 p messages. The code is logical that someone with basic programming skill could follow along and finish out the programming. Interested? Thanks, Ryan From: wo...@justfamily.org Date: Wed, 26 Nov 2014 09:06:53 -0700 To: cisco-voip@puck.nether.net Subject: [cisco-voip] IMP external database front-end Before I go down the road of re-creating the wheel, has anyone written a nice front end to the postgresql database for simple reports for a customer? While pgadmin will do the trick, a non-technical person will struggle with getting reports out as needed. I'm thinking about a simple search for person(s), date/time range, probably PHP would be easiest since if you installed the phppgadmin on a linux install, you have php and apache already there, or maybe on a 2nd box for management purposes if it's a big install. I'm not a coder, but I can probably hack something together, would be a lot easier if someone else has something to share :) Thanks!Happy and safe Holidays to everyoneCharles ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Jabber contact disappears consistently
I call BS! I am running a fully AD integrated 9.1.2 CUCM with IMPresence 9.1.2 and I have the Cisco Jabber for Windows 10.5 client deployed and I have URI Dialing running in the Jabber client, ALL. DAY. LONG and it works flawlessly. In the deployment docs, it says to use EnableSIPUriDialing in the jabber-config.xml HOWEVER; if you look at the PRT logs generated from one of the 10.5 Jabber clients I bet you it is looking for the EnableSIPURIDialing node and not the EnableSIPUriDialing node. Notice the casing difference in URI Vs. Uri . I couldn't get URI in the client for weeks, got the same junk answer from TAC as well then I troll'ed the PRT logs and found that to be my issue, a flippin' casing issue. Date: Wed, 26 Nov 2014 09:09:50 -0600 From: pav.c...@gmail.com To: leslie.me...@lvs1.com CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Jabber contact disappears consistently Just to close this out. Official word from Cisco even though every piece of Jabber doc says its supported in J4W 10.5 == Currently Directory URI is only supported on the server. The client has not yet been updated to fully support this. So no, Jabber 10.5 does NOT support directory URI == On Sat, Nov 22, 2014 at 2:15 PM, Leslie Meade leslie.me...@lvs1.com wrote: I also have seen the same issue, and put everyone onto the same server Leslie Meade .. Mobile:778.228.4339 | Main: 604.676.5239 Email: leslie.me...@lvs1.com From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason Aarons (AM) Sent: Saturday, November 22, 2014 11:33 AM To: Josh Warcop; Pavan K; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Jabber contact disappears consistently I complete agree, put all Jabberusers on single server. Saw the same bugs in 10x. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Josh Warcop Sent: Friday, November 21, 2014 3:48 PM To: Pavan K; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Jabber contact disappears consistently Do to the bugs I've seen I have been pinning all users to a single node in the cluster and just having HA failover. Date: Fri, 21 Nov 2014 11:31:39 -0600 From: pav.c...@gmail.com To: cisco-voip@puck.nether.net Subject: [cisco-voip] Jabber contact disappears consistently We have an interesting problem on a new jabber deployment that has us stumped. Wonder if anybody else saw this. Two jabber servers with ha enabled and balanced users. Using 10.5su1 for jabber windows and ucm/imp. Leveraging sip directory uri as the IM scheme due to a multi forest environment with duplicate Samaccountnames across domains. UserA contact list has userB on it. Folks can im each other without any problem. If we move userB from his imp server to another server in the same subcluster, userB disappears from userA's contact list. Repeatable across multiple users with different userA and userB and happens every time regardless of moving them from server1 to server2 or vice versa. Using router to router communication between nodes and default jabber-config. Any ideas ? Have a TAC case open but its going nowhere. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip itevomcid -- - Pavan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Extension Mobility login is unavailable (23)
Is the phone subscribed to the Extension Mobility Service? Is Extension Mobility capabilities enabled on the phone that the user is trying to log into? From: leslie.me...@lvs1.com To: cisco-voip@puck.nether.net Date: Thu, 27 Nov 2014 16:21:59 + Subject: [cisco-voip] Extension Mobility login is unavailable (23) Client is getting this error when they try to use EM. The error tells me that it is to do with EMCC and they are not configured for this, any ideas ? I have restarted the EM service in the cluster and it was working before hand. Cheers Leslie ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Extension Mobility login is unavailable (23)
Any database replication issues at the moment? Are the phones still able to resolve the address of the TFTP server at the moment (either by IP address or hostname) Is the URL that you are specifying in the Extension Mobility Service still valid and resolvable by the phones? Thanks, Ryan From: leslie.me...@lvs1.com To: ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Extension Mobility login is unavailable (23) Date: Thu, 27 Nov 2014 16:31:57 + Yes to all. It is not just one site. I have tried other phones in the cluster and it is also happening to them as well. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Thursday, November 27, 2014 8:31 AM To: Leslie Meade; cisco-voip cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Extension Mobility login is unavailable (23) Is the phone subscribed to the Extension Mobility Service? Is Extension Mobility capabilities enabled on the phone that the user is trying to log into? From: leslie.me...@lvs1.com To: cisco-voip@puck.nether.net Date: Thu, 27 Nov 2014 16:21:59 + Subject: [cisco-voip] Extension Mobility login is unavailable (23) Client is getting this error when they try to use EM. The error tells me that it is to do with EMCC and they are not configured for this, any ideas ? I have restarted the EM service in the cluster and it was working before hand. Cheers Leslie ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Cisco Presence External Database Front End Application
I have had a few requests from folks for help with a front-end GUI application for the external PostgreSQL database that can be used with the Cisco Presence and IM server. Using something like PgAdmin is a great tool to use in an administrative function but not very user friendly for the non tech. I have written a PHP based application (developed on a LAMP stack). That is a great front-end GUI for basic functionality. The application has 2 of the 4 features finished and can be a great learning tool or a head start to writing/expanding your own app. Please download at: http://ryanthomashuff.com/downloads/ Let me know if you have any questions/need assistance getting it set up. Thanks, Ryan Huff ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Jabber 10.5.1 for Windows shows In a Meeting when he isn't
Is this local client integration or is the CPIM server integrated with an on-prem Exchange server? Thanks, Ryan Huff ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Tue, 2 Dec 2014 16:31:49 + Subject: [cisco-voip] Jabber 10.5.1 for Windows shows In a Meeting when he isn't We have a user who’s Jabber for Windows 10.5.1 client shows “In a Meeting” when his calendar is clear. I looked in outlook if he has multiple calendars which he did not, when he logs into Jabber his status shows as “available” for around 15 seconds before changing to “In a meeting”. If I clear the local Jabber CSF folders it still displays the same behavior. IMP 10.5.1 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 7941 SIP Endpoints Rejected - CUCM 10.5
Is the phone's config pre-built on CCM or is the phone trying to auto register? Thanks, Ryan Huff ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist Date: Thu, 4 Dec 2014 20:49:50 -0500 From: joel.dav...@gmail.com To: cisco-voip@puck.nether.net Subject: [cisco-voip] 7941 SIP Endpoints Rejected - CUCM 10.5 Trying to get some 7941 handsets registered as SIP endpoints for a customer that is coming back via a s2s IPSEC VPN tunnel. We are able to see the traffic come across and the validated it did upgrade but the current phone we are testing with just shows as rejected. Besides pulling a pcap on the subscriber/tftp nodes anyone else run into this or seen similar issues? There is firewall traversal but SIP ALG and protocol inspection has been disabled. -- Joel Davila ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Replace 7940 with 7942
Add a new phone device using the 7942 device type. Then build out the device config line for line compared with the 7940 config. Pay attention the phone template/softkey templates ... etc. Thanks, Ryan Huff ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist Date: Wed, 17 Dec 2014 09:10:56 -0500 From: dzh...@gmail.com To: cisco-voip@puck.nether.net Subject: [cisco-voip] Replace 7940 with 7942 I have a two line 7940 setup but the internal mic is not working. SO i bought a 7942 replacement. If I were doing a 7940 for 7940 replacement, I would just login to CM to do a MAC Address change. What is the best/easiest way to do this since I am replacing it with a 7942? Thanks. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Call track
Are these recent calls or historical? Do you currently have any type of call accounting system where your CDR's are currently sent? If these are recent calls you can start with the Real Time Data in the RTMT. Thanks, Ryan Huff ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist Date: Fri, 19 Dec 2014 00:12:32 +0530 From: dharambi...@gmail.com To: cisco-voip@puck.nether.net Subject: [cisco-voip] Call track Hi Guys, I am using CUCM 7 cluster and with multi VG MGCP at different location. My Telecom provider has given list of unauthorized calls made from us. So where we can investigate in cucm.I collected CDR a nd cucm trace thert are showing some unsuccessful call attempts. Is there any strong methodology to collect all reports -- Regards, Dharambir Kumar ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] 10.5 Prime License Manager HA Deployment
Looking to deploy two PLM servers in an HA fashion. I assume that I just need to add a reference to the second PLM in the CCM Server section like a typical CCM subscriber. Is that correct? Thanks, ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] 10.5 BAT Error/Question
I think I know what the issue is but can't seem to verify ... - 4 node 10.5 Cluster with DNS enabled. - Currently, I have a known DNS resolution problem in the cluster, I know that and am working on it. What currently happens is I'll upload a file in BAT then run a job (like import) against the file I uploaded and the Job Scheduler log comes back with a 'success' result but N/A items processed. When I open the log I get Error; cannot reference /./filename.tar. I'm guessing that BAT is using the DNS resolver to reference node/path/file and since DNS is bunk right now, it can't resolve the node. I can't find in the Googles and docs where it definitively says BAT uses DNS when enabled. Is that my issue or should I be looking elsewhere? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Device pack installation methodology question
I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support for some 88xx phones but I do not want to update the loads for anything else. The approach I am going to use is: Drop the publisher out of the CM Group, forcing all phones to the subscriber. Install the device pack on the publisher and reboot the publisher. Once the publisher is backup, set all the device defaults back to what I want them to be then add the publisher back to the CM Group. Then drop the subscriber from the CM Group forcing all the phones on the publisher and start the install process over for the subscriber. Once everything is back up add the subscriber back to the CM Group. Does that sound reasonable or is there an easier way? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Device pack installation methodology question
Correct, ordinarily that would be what I would do as well. In this case though, I am trying to avoid upgrading the firmware on anything, other than the devices I am trying to get support for. Thanks, Ryan Date: Sat, 3 Jan 2015 14:20:43 -0600 From: b...@brezworks.com To: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Device pack installation methodology question On 1/3/2015 2:11 PM, Ryan Huff wrote: I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support for some 88xx phones but I do not want to update the loads for anything else. The approach I am going to use is: Drop the publisher out of the CM Group, forcing all phones to the subscriber. Install the device pack on the publisher and reboot the publisher. Once the publisher is backup, set all the device defaults back to what I want them to be then add the publisher back to the CM Group. Then drop the subscriber from the CM Group forcing all the phones on the publisher and start the install process over for the subscriber. Once everything is back up add the subscriber back to the CM Group. Does that sound reasonable or is there an easier way? Any time I've needed device support I just install it on all the nodes then reboot them one at a time during a maintenance window. As long as both nodes are running call processing, they'll fail over when the node reboots. Not sure about on 9.X, but on 10.X this just shows as a blip on the phones of them reregistering, not a full reboot. Obviously if you install new software for those phones, they'll upgrade software when they reboot. Jeremy TheBrez Bresley b...@brezworks.com ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCMC 10.5.1 to CXN 10.5.1 with SCCP number of rings 4 before answer
Is it one phone or all phones? It sounds like an interdigit timeout or extended hunt; I know you said there are no overlaps though. Could be on the connections side too if it is system wide. Anything change with the CSS/Partion of the ports? Distribution algorithm of the line group change? I assume all the SCCP ports are still registered. What does RTMT say about connection's health, CPU spike? Thanks, Ryan From: jason.aar...@dimensiondata.com To: cisco-voip@puck.nether.net Date: Mon, 2 Feb 2015 17:52:56 + Subject: [cisco-voip] CUCMC 10.5.1 to CXN 10.5.1 with SCCP number of rings 4 before answer Customer states that it’s taking too long for Unity Connection to answer when he presses the Messages button. It’s SCCP integrated. The Hunt Pilot looks good, no duplicate or unassigned DNs, no overlapping number range. Been in production and was working fine previously. I did notice the checkbox in the Hunt List “For Voice Mail Usage” is unchecked. After hours we can check that. CUCMC 10.5.1.1-7 CXN 10.5.1.11900-13 Jason Aarons Consultant Dimension Data 904-338-3245 mobile ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Web directory from CUCM
Terry, I am literally moving to Georgia today and tomorrow! I'll save this email look for something from me early next week. It's all my code so I can and will share, no problem. Thanks, Ryan Huff___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Web directory from CUCM
I have always made mine in the past. I use PHP for my scripting needs. If you have a way to run PHP, I can share - just ping me. Thanks, Ryan Huff___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] A new web GUI for Cisco Call Manager
Hello all, I thought I'd share a PHP based web GUI for the Corporate Directory feature of Cisco Call Manager. I made it a while ago and have seen a few recent requests for something like it, so I thought I put it out here. There is a README in the ZIP archive with all the particulars. It is a basic design with a little CSS (Cascading Style Sheet) so you'll probably want to change the interface around a little to fit your needs, but all the backend code is solid. http://ryanthomashuff.com/2015/01/new-web-gui-for-call-manager-corporate-directory/ Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Unity connection - Voicemail playback
Jose, If SIP integrated, easy fix, just point the trunk at the pub (and then reset the trunk in CUCM). If SCCP integrated, you have a few different options and the best option depends on how you're setup and if this is a perm. change or just temporary? If this is just temporary and assuming you have the max amount of ports per CUC node, In CUCM I would go to the line group of the SCCP ports and drop out the ports that are registered to the CUC subscriber node. Once your link is no longer saturated, add the ports back to the line group. There are some other design strategies that you can look at to help mitigate this in the future. For example, you could add an additional CUC Subscriber node in the same DC as the publisher (so you end up with a subscriber in both data centers). This would allow you to have additional ports in the same DC as the publisher. Thanks, Ryan From: chrw...@cisco.com To: jcolon...@gmail.com; cisco-voip@puck.nether.net Date: Mon, 16 Feb 2015 18:52:43 + Subject: Re: [cisco-voip] Unity connection - Voicemail playback You have to route calls to that node. Right now, either you SCCP ports, or SIP trunk are pointing to the Sub. You have to change the priority of the trunk or the ports so it uses the Sub first. +Chris TME - Unity Connection and MediaSense From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jose Colon II Sent: Monday, February 16, 2015 1:44 PM To: Cisco VOIP Subject: [cisco-voip] Unity connection - Voicemail playback I have a Unity 10.5 HA setup and the subscriber is at our COLO. Currently the link between the two is being saturated so voicemail playback is very choppy. The question is, how do i force the publisher to handle those requests so that voicemail playback is handled inside this location and not at the colo? Thanks Jose ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CCNA/CCNP Collaboration certs
Dennis, The conversion tool shows that CIPTV2 is the only test that CCNP Voice needs to convert to CCNP Collaboration. This is really new info though so it could change. As of this writing the CCNP Voice retirement date is listed as Q4 2015 and PearsonVue doesn't even have 300-075 (CIPTV2) listed as a schedue-lable test yet. From what I understand, the CIPTV2 test covers video related material. Given that the CCIE Collaboration is based on the 9.x stack I assume the topics will cover things like Collaboration Edge, Jabber Video, Telepresence ... etc Thanks, Ryan From: dennis.h...@wwt.com To: ryanh...@outlook.com; ben.st...@gmail.com; wo...@justfamily.org CC: cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CCNA/CCNP Collaboration certs Date: Tue, 17 Feb 2015 18:50:05 + What exam is required to convert CCNP voice to CCNP collaboration? Dennis Heim | Emerging Technology Architect (Collaboration) World Wide Technology, Inc. | +1 314-212-1814 Innovation happens on project squared -- http://www.projectsquared.com Click here to join me in my Collaboration Meeting Room From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: Tuesday, February 17, 2015 11:58 AM To: Ben Story; Charles Goldsmith Cc: voip puck Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs What gets me is that a CCNP Voice needs one test to upgrade to CCNP Collaboration (http://www.cisco.com/web/learning/tools/ccnp_collab/ccnp_collab_tool.html). Really? Only one test's worth of info is what changes between Voice and Collaboration? If you are going to retire and invalidate a credential and move to another tract altogether, then it should be a top-down revamp, not just one test of a few new questions about video. This feels like it is nothing more than re branding/marketing; and I shouldn't have to pay for that. My grumpy two cents, Ryan Date: Tue, 17 Feb 2015 10:10:24 -0600 From: ben.st...@gmail.com To: wo...@justfamily.org CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs Is it just me or does it seem weird for a CCNA level to now require two tests. Money grab? -- Ben Story CCNP, CCNA, CCNA Wireless, CCDA ben.st...@gmail.com @ntwrk80 http://showbrain.blogspot.com http://rand0mw0rds.blogspot.com From sour-faced saints and silly devotions, good Lord, preserve us!. -- St. Teresa of Avila On Tue, Feb 17, 2015 at 9:58 AM, Charles Goldsmith wo...@justfamily.org wrote: https://learningnetwork.cisco.com/community/ccna-ccnp-collaboration ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CCNA/CCNP Collaboration certs
What gets me is that a CCNP Voice needs one test to upgrade to CCNP Collaboration (http://www.cisco.com/web/learning/tools/ccnp_collab/ccnp_collab_tool.html). Really? Only one test's worth of info is what changes between Voice and Collaboration? If you are going to retire and invalidate a credential and move to another tract altogether, then it should be a top-down revamp, not just one test of a few new questions about video. This feels like it is nothing more than re branding/marketing; and I shouldn't have to pay for that. My grumpy two cents, Ryan Date: Tue, 17 Feb 2015 10:10:24 -0600 From: ben.st...@gmail.com To: wo...@justfamily.org CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs Is it just me or does it seem weird for a CCNA level to now require two tests. Money grab?--Ben Story CCNP, CCNA, CCNA Wireless, ccdaben.st...@gmail.com @ntwrk80http://showbrain.blogspot.comhttp://rand0mw0rds.blogspot.com From sour-faced saints and silly devotions, good Lord, preserve us!. -- St. Teresa of Avila On Tue, Feb 17, 2015 at 9:58 AM, Charles Goldsmith wo...@justfamily.org wrote: https://learningnetwork.cisco.com/community/ccna-ccnp-collaboration ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...
CUCM 10.5 (New Build with DNS) Testing out phone registration with IP Communicator 8.6.4. Phone auto registers just fine and is assigned all the auto registered defaults I have specified (DN, partition, CSS ... etc) and the IP communicator is able to resolve all the URLs by DNS. Everything appears to be fine. However, the Services menu keeps poping up. I haven't configured any services beyond the shipped services so the service menu is blank. I hit the exit button on the services menu and it goes away for a second or two, then it comes right back. The phone can still process digits, I can dial, I can get dialtone ... etc everything seems to work but the services menu. I feel like I have dealt with this before but I can't recall what the solution was. Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...
Winner winner chicken dinner! Thank you, that was driving me nuts. Learning all the new things in 10 ... coming from the 8/9 world doesn't seem that long ago but in Cisco UC land I might as well be a dinosaur. Thanks, Ryan Date: Thu, 29 Jan 2015 11:03:22 -0600 Subject: Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ... From: ad...@adman.net To: ryanh...@outlook.com CC: cisco-voip@puck.nether.net The default Universal Device Template includes an idle URL that tries to access the self-provisioning IP Phone Service (/cucm-uds/xps/selfProvision). I'm guessing this is what you're seeing. On Thu, Jan 29, 2015 at 10:52 AM, Ryan Huff ryanh...@outlook.com wrote: CUCM 10.5 (New Build with DNS) Testing out phone registration with IP Communicator 8.6.4. Phone auto registers just fine and is assigned all the auto registered defaults I have specified (DN, partition, CSS ... etc) and the IP communicator is able to resolve all the URLs by DNS. Everything appears to be fine. However, the Services menu keeps poping up. I haven't configured any services beyond the shipped services so the service menu is blank. I hit the exit button on the services menu and it goes away for a second or two, then it comes right back. The phone can still process digits, I can dial, I can get dialtone ... etc everything seems to work but the services menu. I feel like I have dealt with this before but I can't recall what the solution was. Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...
So I just changed the default phone template in the Cisco IP Communicator Device Default to solve this particular issue. Thanks for the bug id reference Brian, that helps! Thanks, Ryan Date: Thu, 29 Jan 2015 12:16:35 -0500 Subject: Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ... From: bmead...@vt.edu To: ad...@adman.net CC: ryanh...@outlook.com; cisco-voip@puck.nether.net Also changing it there will remove it for new IP Phones added but you need to go into any phone that already auto-registered and remove the idle URL on the device configuration. This feature only works on phones that support SBD/TVS since it works over HTTPS only. That's why it doesn't work with IP communicator. I opened a doc bug on this- https://tools.cisco.com/bugsearch/bug/CSCun13382 Brian On Thu, Jan 29, 2015 at 12:03 PM, Adam Blomfield ad...@adman.net wrote: The default Universal Device Template includes an idle URL that tries to access the self-provisioning IP Phone Service (/cucm-uds/xps/selfProvision). I'm guessing this is what you're seeing. On Thu, Jan 29, 2015 at 10:52 AM, Ryan Huff ryanh...@outlook.com wrote: CUCM 10.5 (New Build with DNS) Testing out phone registration with IP Communicator 8.6.4. Phone auto registers just fine and is assigned all the auto registered defaults I have specified (DN, partition, CSS ... etc) and the IP communicator is able to resolve all the URLs by DNS. Everything appears to be fine. However, the Services menu keeps poping up. I haven't configured any services beyond the shipped services so the service menu is blank. I hit the exit button on the services menu and it goes away for a second or two, then it comes right back. The phone can still process digits, I can dial, I can get dialtone ... etc everything seems to work but the services menu. I feel like I have dealt with this before but I can't recall what the solution was. Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] BE7K Unity Connections 5,000 User
Looks like the DocWiki says I can run Unity Connections 5,000 user on the BE7K-K9 SKU but states an OVA mod needed for 8.x and 9.x. I am guessing the same holds true in 10.5? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Device pack installation methodology question
I like that approach - seems simpler, does mean running the phones without valid tftp for a few (ringtones, phone desktops, corp. directory ...) but that's what maintenance windows are for. Thanks for your time! Thanks, Ryan From: avholloway+cisco-v...@gmail.com Date: Sat, 3 Jan 2015 20:58:09 + Subject: Re: [cisco-voip] Device pack installation methodology question To: ryanh...@outlook.com; cisco-voip@puck.nether.net I just did this recently too, to add support for an 8831 on an 8x cluster. Trying to recall from memory, here's what I recall doing: 1. Deactivate TFTP on both nodes2. BAT Export the Device Defaults3. Install the Dev Pack on Pub4. Restart Pub5. Install Dev Pack on Sub6. Restart Sub7. BAT Import the Device Defaults8. Activate TFTP on both nodes9. Add new 8831's For step 7, I also did a BAT export of device defaults post dev pack, and ran a diff on the two. I saw the changes to the existing phone firmware as well as the addition of the new phone models. When you import the old device defaults back in at this stage, note that the absence of the new phone models simply tells CUCM BAT process to ignore them and leave them alone, while reverting the firmware changes on all of the existing phones. On Sat Jan 03 2015 at 2:15:25 PM Ryan Huff ryanh...@outlook.com wrote: I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support for some 88xx phones but I do not want to update the loads for anything else. The approach I am going to use is: Drop the publisher out of the CM Group, forcing all phones to the subscriber. Install the device pack on the publisher and reboot the publisher. Once the publisher is backup, set all the device defaults back to what I want them to be then add the publisher back to the CM Group. Then drop the subscriber from the CM Group forcing all the phones on the publisher and start the install process over for the subscriber. Once everything is back up add the subscriber back to the CM Group. Does that sound reasonable or is there an easier way? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Where is the Unified Attendant Console ISO download?
Hello, I can find the OVA to download in the portal but I can't seem to find the actual ISO. Any clues? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Quick question about Unified Attendant Console
Not really familiar with installing U-AC from what I am gathering (version 10.5); it is still Windows dependant, correct? Use the OVA to create the machine that you install Windows on, then use the AC binary to install. Is that right? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Advanced Attendant Console 10.5, which Win OS?
Anyone know where I can find which Win OS is required for the Advanced Attendant Console Server 10.5? Thanks, Ryan Huff r...@ryanthomashuff.com http://www.ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNA Voice CCNP Voice, CCIE Collaboration (Written) UCCX Specialist ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Intercluster Trunks
How are you doing your media termination? Which side of them trunk is terminating? Thanks, Ryan Huff___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Unity Connections - COBRAS import and a question
So I have no issue using COBRAS to import vm accounts, call handlers and such ... How about some of the more obscure elements though, like restriction tables, COS ... etc. Does all that need to be hand rebuilt from the MS Access backup or is their an easier way that I'm just missing Thanks, Ryan Huff ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCILYNC Voicemail login issues
Another thing to look at / consider is client DNS issues (if the client is trying to resolve the directory server by hostname). Which, should be evident in the p-caps. Thanks, Ryan From: jason.aar...@dimensiondata.com To: ryanh...@outlook.com; george.hend...@l-3com.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues Date: Tue, 13 Jan 2015 14:46:50 + Capture the packets via Wireshark/network. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: Tuesday, January 13, 2015 9:19 AM To: george.hend...@l-3com.com; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUCILYNC Voicemail login issues Have you reconfirmed all your integration credentials? You might not get a directory error because the bind request might not even be making it to your directory server. Did anything change about the directory server (ip address, bind account credentials, password expire ... etc)? Can the clients still talk to the directory server over the required ports? Thanks, Ryan From: george.hend...@l-3com.com To: cisco-voip@puck.nether.net Date: Tue, 13 Jan 2015 14:03:18 + Subject: [cisco-voip] CUCILYNC Voicemail login issues Hey Guys, I’m seeing an issue where I am getting “invalid username or password” when trying to authenticate in CUCILYNC for Voicemail, but I don’t get an error for the Directory services. However, I can login to the Cisco PCA page just fine with my credentials. Nothing has changed with the voicemail system and all the registry entries are correct for pointing to the voicemail system for CUCILYNC. Any ideas what could cause this? Thanks, Bill ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip itevomcid ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCILYNC Voicemail login issues
You can have it either way (depending on your configuration), whatever is being advertised to your client for directory auth is what your client is trying to contact. I would fire up wireshark as suggested and see what is going on in packet land, could just be a silly network issue. Thanks, Ryan From: george.hend...@l-3com.com To: ryanh...@outlook.com; jason.aar...@dimensiondata.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues Date: Tue, 13 Jan 2015 14:54:12 + Does the CUCILYNC client go straight to LDAP server (the one set in the registry) for Voicemail authentication or to the voicemail server? Service account for Unity Connection in AD is fine. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Tuesday, January 13, 2015 9:51 AM To: Jason Aarons AM; Hendrix, George (Bill) @ NSS; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues Another thing to look at / consider is client DNS issues (if the client is trying to resolve the directory server by hostname). Which, should be evident in the p-caps. Thanks, Ryan From: jason.aar...@dimensiondata.com To: ryanh...@outlook.com; george.hend...@l-3com.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues Date: Tue, 13 Jan 2015 14:46:50 + Capture the packets via Wireshark/network. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: Tuesday, January 13, 2015 9:19 AM To: george.hend...@l-3com.com; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUCILYNC Voicemail login issues Have you reconfirmed all your integration credentials? You might not get a directory error because the bind request might not even be making it to your directory server. Did anything change about the directory server (ip address, bind account credentials, password expire ... etc)? Can the clients still talk to the directory server over the required ports? Thanks, Ryan From: george.hend...@l-3com.com To: cisco-voip@puck.nether.net Date: Tue, 13 Jan 2015 14:03:18 + Subject: [cisco-voip] CUCILYNC Voicemail login issues Hey Guys, I’m seeing an issue where I am getting “invalid username or password” when trying to authenticate in CUCILYNC for Voicemail, but I don’t get an error for the Directory services. However, I can login to the Cisco PCA page just fine with my credentials. Nothing has changed with the voicemail system and all the registry entries are correct for pointing to the voicemail system for CUCILYNC. Any ideas what could cause this? Thanks, Bill ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip itevomcid ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Cisco Unity Connections 10.5 Intersite Networking Methodology
Can someone pass me some me some links/info on Intersite/Intrasite networking? I know what they are are ... etc but I have still have some methodology/strategy questions. Like why would I consider using it? What advantages do I get by linking to CUC clusters together , what can I do with it ... etc Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] LDAP Search Filter Syntax reference
Does anyone have any links / info for LDAP Search Syntax? I am trying to write this filter: Sync all children in ABC OUExcept for children in XYZ OUANDAll children of ABC OU WHERE IpPhone IS NOT null Thanks, Ryan Huff r...@ryanthomashuff.com http://www.ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNA Voice CCNP Voice, CCIE Collaboration (Written) UCCX Specialist ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Strange routing behavior cucm 10.5 and int'l pattern
Try this one on; Was working fine ... Standard Int'l route pattern 9.011!# Discard set to PreDot (again, this was working ... no issue on gateway ... etc) So today it stops working, just rings busy. I debug the ISDN and it shows called party as the last 7 digits. I go over to DNA and use an int'l pattern with the css I was using and it blocks pattern for unallocation. I create a new test partition with a new 9.011!# pattern in it and a new css with only the new partition in it. I go back over to DNA and try the int'l pattern with the new test css and it blocks for unallocated. Now I scratch my head, so I take off the octothorpe on the pattern (9.011!) and BOOM, DNA routes and everything is happy. I move to production and it works just fine without the octothorpe. What does this sound like? Do you think I may have competing patterns somewhere in the dial plan? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Strange routing behavior cucm 10.5 and int'l pattern
Andrew, Since international numbers are varying lengths, the # is used to signal the end of the number string. You can also just wait for the interdigit timeout to expire and then routing occurs. Usually as Brian mentioned, both are supported because it usually comes down to a user preference. Thanks, Ryan Huff___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Strange routing behavior cucm 10.5 and int'l pattern
Brian, I suppose I wasn't clear on that. The octothorpe was dialed at the end of the international pattern in production and with DNA. Only when I removed the octothorpe from the pattern did routing occur. Thanks, Ryan Huff___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CallerID not showing on 8941
What version of CCM Leslie? If you're on 8.x+ You could try to create a 'Route next hop by calling party number translation and prefix the + in the on the xlate pattern with the anticipated CLID and see if you can catch it. If you prove that it is the + then you can translate/transform and remove. Again, this would be a stop-gap work around. Thanks, Ryan Huff r...@ryanthomashuff.com http://www.ryanthomashuff.com CCNA R/S, CCNA Wireless, CCNA Voice CCNP Voice, CCIE Collaboration (Written) UCCX Specialist Date: Thu, 8 Jan 2015 11:24:04 -0500 From: bmead...@vt.edu To: leslie.me...@lvs1.com CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CallerID not showing on 8941 Probably has to do with the special character. Here was the only bug I found related- https://tools.cisco.com/bugsearch/bug/CSCuo91983 Might be caused by the same root issue but might need a new bug opened. On Thu, Jan 8, 2015 at 11:12 AM, Leslie Meade leslie.me...@lvs1.com wrote: CallerID from the PSTN to H323 is correct. Sip trunk to CUCM is correct. CUCM to 8841 shows correct. But is it not displaying on the screen. Same call to another model is fine. I suspect the issue lies with the “±” in front of the Displayname ? 16301690.001 |10:27:08.571 |AppInfo |localizeCgpn: StationSIPCdfc on device SEP6CFA89720901 , CSS = ,useDevicePoolCgpnCss =1 AlternateCgpn(global)=705264 cgpn=705264 16301690.002 |10:27:08.571 |AppInfo |SIPStationCdfc:CcSetupReq - unicodeConnectedUnicodeDisplayName='' asciiConnectedDisplayName='±TRUE NORTH TRAN' 16301690.003 |10:27:08.571 |AppInfo |SIPStationCdfc:CcSetupReq - unicodeCallingPartyName='' asciiCallingPartyName='±TRUE NORTH TRAN' callingParty='705264' Has anyone seen something like this before ? Leslie ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Transcoding question
Thanks Dainel! I probably knew that at some point, but I couldn't remember for the life of me! Makes total sense. Thanks, Ryan Original Message From: Daniel Pagan dpa...@fidelus.com Sent: Monday, March 16, 2015 09:16 PM To: Daniel Pagan dpa...@fidelus.com,Ryan Huff ryanh...@outlook.com,cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Transcoding question For clarity, by “higher bandwidth codec” I meant to say higher bit-rate codec, or codec of higher bandwidth consumption. - Dan From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Daniel Pagan Sent: Monday, March 16, 2015 9:08 PM To: Ryan Huff; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Transcoding question Hey Ryan how’s it going? Transcoder allocated by CUCM comes from the side using a higher bandwidth codec, regardless if it’s the calling or called party, with the intention to avoid streaming a high bandwidth consuming codec over a WAN connection – keeping it local to the LAN. Of course, this isn’t always true, such as due to a local transcoding resource being entirely nonexistent or a misconfiguration of the MRG/MRGLs. Hope this helps answer your question. - Dan From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: Monday, March 16, 2015 8:11 PM To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: [cisco-voip] Transcoding question When xcoding is required in the call setup, which side is transcoded? The called party or the calling party? Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Transcoding question
When xcoding is required in the call setup, which side is transcoded? The called party or the calling party? Thanks___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Extension that hangs up on the user?
So if the CUC AA IS NOT playing a greeting and the AA is doing NOTHING but the after action of hang-up; you could just create a translation pattern that matches the called number (presumably, the called number is currently a CTI route point/DN that is forwarding to CUC, you would need to remove it before creating the translation). In the translation pattern, set block this pattern, call rejected. You can also explore CCM ANI based call blocking; I discuss it here. http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ Thanks, Ryan From: jandrewar...@ccgs.wa.edu.au To: ryanh...@outlook.com; cisco-voip@puck.nether.net Date: Wed, 18 Mar 2015 22:10:36 +0800 Subject: RE: [cisco-voip] Extension that hangs up on the user? Yeah, CUC has Callers Hear: Nothing then After Greeting/Call Action: Hang Up. How would I dump the call? Select Route Option/Block this pattern: Call Rejected in the translation pattern? Thanks, James Andrewartha Network Projects Engineer Christ Church Grammar School Claremont, Western Australia Ph. (08) 9442 1757 Mob. 0424 160 877 From: Ryan Huff [ryanh...@outlook.com] Sent: Wednesday, 18 March 2015 8:12 PM To: James Andrewartha; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Extension that hangs up on the user? So is CUC just so you can use the after action hang up technique or are you playing a greeting first? If your just hanging up and not playing a greeting, could you just catch the ingress call on a translation then just dump the call (or play the reorder tone)? Thanks, Ryan Original Message From: James Andrewartha jandrewar...@ccgs.wa.edu.au Sent: Wednesday, March 18, 2015 02:34 AM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Extension that hangs up on the user? Hi list, Is there a way in CUCM to make an extension that hangs up on the other end? Currently we have a Unity Connection AA that does that, but it's literally the only thing CUC is being used for and I want to get rid of it. Currently we have our AAs (and voicemail) in Exchange 2007, which is being upgraded to 2013 soon, but as far as I can tell there's no way to have it hang up on the caller, so I transfer to the AA in Unity. Thanks, -- James Andrewartha Network Projects Engineer Christ Church Grammar School Claremont, Western Australia Ph. (08) 9442 1757 Mob. 0424 160 877 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Extension that hangs up on the user?
Very true. Just leave it unallocated and CCM doesn't have to do anything, but the caller will get annunciation. Thanks, Ryan Original Message From: Daniel Pagan dpa...@fidelus.com Sent: Wednesday, March 18, 2015 11:17 AM To: Ryan Huff ryanh...@outlook.com,James Andrewartha jandrewar...@ccgs.wa.edu.au,cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Extension that hangs up on the user? The main question I have... if CUC is being used simply to hang-up on the calling party, what's the purpose of needing this migrated to CUCM instead of simply leaving the number unallocated? Correct me if I'm wrong, but it seems to me that you're specifically looking for a method in CUCM where the call is answered and then disconnected. Is this true? Are you hoping to have the call actually connected before the disconnect? Or does a simple rejection of the call work fine for you? Only situation I can think of where this is needed would be not wanting callers to hear a rejection or error recording due to unallocated number. - Dan From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: Wednesday, March 18, 2015 10:23 AM To: James Andrewartha; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Extension that hangs up on the user? So if the CUC AA IS NOT playing a greeting and the AA is doing NOTHING but the after action of hang-up; you could just create a translation pattern that matches the called number (presumably, the called number is currently a CTI route point/DN that is forwarding to CUC, you would need to remove it before creating the translation). In the translation pattern, set block this pattern, call rejected. You can also explore CCM ANI based call blocking; I discuss it here. http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ Thanks, Ryan From: jandrewar...@ccgs.wa.edu.aumailto:jandrewar...@ccgs.wa.edu.au To: ryanh...@outlook.commailto:ryanh...@outlook.com; cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Date: Wed, 18 Mar 2015 22:10:36 +0800 Subject: RE: [cisco-voip] Extension that hangs up on the user? Yeah, CUC has Callers Hear: Nothing then After Greeting/Call Action: Hang Up. How would I dump the call? Select Route Option/Block this pattern: Call Rejected in the translation pattern? Thanks, James Andrewartha Network Projects Engineer Christ Church Grammar School Claremont, Western Australia Ph. (08) 9442 1757 Mob. 0424 160 877 From: Ryan Huff [ryanh...@outlook.com] Sent: Wednesday, 18 March 2015 8:12 PM To: James Andrewartha; cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Extension that hangs up on the user? So is CUC just so you can use the after action hang up technique or are you playing a greeting first? If your just hanging up and not playing a greeting, could you just catch the ingress call on a translation then just dump the call (or play the reorder tone)? Thanks, Ryan Original Message From: James Andrewartha jandrewar...@ccgs.wa.edu.aumailto:jandrewar...@ccgs.wa.edu.au Sent: Wednesday, March 18, 2015 02:34 AM To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: [cisco-voip] Extension that hangs up on the user? Hi list, Is there a way in CUCM to make an extension that hangs up on the other end? Currently we have a Unity Connection AA that does that, but it's literally the only thing CUC is being used for and I want to get rid of it. Currently we have our AAs (and voicemail) in Exchange 2007, which is being upgraded to 2013 soon, but as far as I can tell there's no way to have it hang up on the caller, so I transfer to the AA in Unity. Thanks, -- James Andrewartha Network Projects Engineer Christ Church Grammar School Claremont, Western Australia Ph. (08) 9442 1757 Mob. 0424 160 877 ___ cisco-voip mailing list cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Migration strategy
I'm not, sure I completely understand your questions but I'll attempt to answer based on my understanding. Yes, you can pre-config devices and users in CCM prior to migration. If they are Cisco IP phones, you'll need the MAC address and model of the phone at a minimum. If they are non Cisco IP phones, you'll need to pre configure 3rd party sip devices (which is a different license requirement than a Cisco IP phone). Place the preconfigured dial plan that isnt migrated yet (on CCM), in a temp. partition that the already migrated phones cannot access. As you migrate, change that partition using BAT, to the correct partition for the portion of phones you migrated. Thanks, Ryan Original Message From: 秀王 kiwi.vo...@gmail.com Sent: Tuesday, March 17, 2015 09:15 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Migration strategy Currently the client have avaya and cisco linked together using SIP. Cisco UCM cluster have users in the production environment. I'm are going to cutover more sites from avaya to cisco. Is it possible to preconfigure the users, extension number (let's say 87XXX range), phones and the user device profiles in advance? I'm thinking that if I preconfigure those information, the cucm will think that those extension number (87XXX) are local and unregistered. Is there a way to make CUCM thinks that in order to reach 87XXX range, it will still reach out to Avaya using the SIP trunk? Is there any setting in the route pattern can do that? I thinking that CUCM will always find a more exact match locally instead of through other source like translation pattern or route pattern. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...
Jonathan, Have you applied any other cop or engineering special files besides the version 3 keys? Is your cluster based on the unrestricted export version or the restricted export version? Does the update your applying match (restricted versus unrestricted) Can you list out the specific version of 10.0.1 you are using and the specific version version of 10.5.1 that you want to go to? If you do a show version active on all the cluster nodes; what do you see? Is it the same on all nodes? I assume you tried to download a new image file from CCO and use that? Thanks, Ryan___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 7841 SIP Phone audio issues on some calls
is the MCB onnet or offnet? Thanks, Ryan Date: Tue, 10 Mar 2015 08:20:16 -0500 From: erick...@gmail.com To: cisco-voip@puck.nether.net Subject: [cisco-voip] 7841 SIP Phone audio issues on some calls Anyone noticing problems on 7841 SIP phones when calling outbound with a SIP provider? When we call Microsoft Conference bridge it connects but we can't hear the bridge. It works fine from a SCCP phone on same Call Manager and same SIP trunk. All other calls work on the 7841 outbound. CUCM Version 10.5.1.1-7 Firmware: sip78xx.10-1-1SR1-4 Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Could you tell me what version of CCM you are dealing with? If you go under Call Routing - Translation Pattern and click Add New, do you see Route Next Hop By Calling Party Number? Thanks, Ryan From: norm.nichol...@kitchener.ca To: cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 12:43:04 + Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 I have been asked to send an outside caller to a voicemail box so when the DNIS or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is this possible ? Thanks Norm Nicholson Telecom Analyst City of Kitchener (519) 741-2200 x 7000 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DNIS Question
Not automatically, I agree.
Thanks,
Ryan
Original Message
From: Daniel Pagan dpa...@fidelus.com
Sent: Friday, March 6, 2015 09:34 AM
To: Ryan Huff
ryanh...@outlook.com,dennis.h...@wwt.com,norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] DNIS Question
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But this doesn’t provide the logic required for the call acceptance rule of
“accept only the first (regardless of ToD), reject all others, reset after 24
hours”. Aside from using some form of UCCX scripting or using a custom app
integrated through the JTAPI or Routing Rules API, I too don’t see CUCM
natively accomplishing this.
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan
Huff
Sent: Friday, March 06, 2015 9:23 AM
To: dennis.h...@wwt.com; norm.nichol...@kitchener.ca;
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DNIS Question
If youre on a modern version of ccm you could use a combo of route to next
hop with calling party I'd and hunts.
That would be a fun one to play around with.
Thanks,
Ryan
Original Message
From: Heim, Dennis dennis.h...@wwt.com
Sent: Friday, March 6, 2015 09:13 AM
To: norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DNIS Question
Out of the box no. You could leverage the Routing Rules API to build a policy
and database lookup.
Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814


Innovation happens on project squared -- http://www.projectsquared.com
Click here to join me in my Collaboration Meeting Room
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
norm.nichol...@kitchener.ca
Sent: Friday, March 06, 2015 6:07 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] DNIS Question
I have a user request for the following:
We are having an issue with a nuisance caller calling dispatch 20-30 times a
day for the past month. Can we set up the phone system to only allow the
following number to call once a day on our and extension numbers ?
519 XXX
After the one call is used for that day, the following call goes to this
message:
“If this is an emergency, please hang up and call 911 (hang up)”
The question is it possible to accept one call a day then route them to a
message or is it all or nothing ?
Thanks
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000
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Re: [cisco-voip] DNIS Question
If youre on a modern version of ccm you could use a combo of route to next hop
with calling party I'd and hunts.
That would be a fun one to play around with.
Thanks,
Ryan
Original Message
From: Heim, Dennis dennis.h...@wwt.com
Sent: Friday, March 6, 2015 09:13 AM
To: norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DNIS Question
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Out of the box no. You could leverage the Routing Rules API to build a policy
and database lookup.
Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814


Innovation happens on project squared -- http://www.projectsquared.com
Click here to join me in my Collaboration Meeting Room
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
norm.nichol...@kitchener.ca
Sent: Friday, March 06, 2015 6:07 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] DNIS Question
I have a user request for the following:
We are having an issue with a nuisance caller calling dispatch 20-30 times a
day for the past month. Can we set up the phone system to only allow the
following number to call once a day on our and extension numbers ?
519 XXX
After the one call is used for that day, the following call goes to this
message:
“If this is an emergency, please hang up and call 911 (hang up)”
The question is it possible to accept one call a day then route them to a
message or is it all or nothing ?
Thanks
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000
___
cisco-voip mailing list
cisco-voip@puck.nether.net
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Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Daniel, There you go trying to make things slick and cool, lol! I like that approach! So, it seems his need is that if only 1 particular DNIS dialed one particular DN (a uccx trigger from my understanding) in ccm, that it then route the call to a specific VM account; but that it would also allow that DNIS to dial any other DN in the cluster without going to that specific VM account.. Being that I love to learn new things constantly; in your approach, is there a way to accomplish that requirement? Thanks, Ryan From: dpa...@fidelus.com To: norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 15:09:32 + Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 For routing based on ANI, Ryan Huff’s suggestion will certainly help from the perspective of CUCM. You can use this alongside CUC routing rules – route calls from the 519 area code to a specific mailbox. If you know these callers from area code 519 are dialing the same DNIS, then routing by ANI in CUCM won’t be needed – simply route the DNIS to voicemail, add a routing rule matching the 519 calls, and route to a mailbox. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of norm.nichol...@kitchener.ca Sent: Tuesday, March 10, 2015 8:43 AM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 I have been asked to send an outside caller to a voicemail box so when the DNIS or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is this possible ? Thanks Norm Nicholson Telecom Analyst City of Kitchener (519) 741-2200 x 7000 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Agreed. CCM ANI call based routing is fantastic, works flawlessly once setup and then you go a month and forget it was setup (because it works so flawlessly) and you make an arbitrary change and BLAMO no ingress in the cluster. Long story short, the CCM approach is great if you're committed to remembering it is there and the role it plays with your dial plan (because it has to insert itself in the ingress path right before the dial plan). Not trying to scare you off of ccm ani routing Norm, it works great and really isn't that hard to setup, but Daniel reminded me of a good point. If you go with the CCM approach, document how you have it setup (names of the partitions and css's ... etc). Thanks, Ryan From: dpa...@fidelus.com To: ryanh...@outlook.com; norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Date: Tue, 10 Mar 2015 17:50:48 + Doc on ANI-based call routing in UCCX: http://www.scribd.com/doc/197424933/Cisco-UCCX-ANI-Based-Call-Routing#scribd Norm can use this as a starting point assuming the call is routing into UCCX already. But use the call redirect step instead of the set step specified in the article. Use call redirect to send the call (which returned TRUE based on the previous IF step looking at calling party) to a DN in CUCM configured for CFwdAll to Unity Connection. Then have a dedicated voice mailbox for these calls with a matching DN. What I’m not sure of is whether CUC will match the voicemail user based on first redirecting number, which might use the CTI port performing the redirect… Not sure… would need to test… but it’s probably the cleanest solution IMO. From: Daniel Pagan Sent: Tuesday, March 10, 2015 1:11 PM To: 'Ryan Huff'; norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 HAH! Slick and cool… Two things I’m far, far from :) Unfortunately it took my email below a full hour to be posted! That’s tricky if the desired behavior is to dial a UCCX trigger and, if the ANI contains area code 519, only then should CUCM route the call to a specific voicemail box. If this is correct, then I think routing based on calling-number in CUCM is certainly going to be required along with dialed number based routing… if scripting isn’t desired... But since it seems they’re dialing a UCCX trigger anyway, assuming it’s not just an unregistered CTI RP and calls are actually routing to UCCX, why not just edit the existing script to read the calling number and redirect the call to a predetermined DN that’s configured for CFWD all into voicemail? Something straight forward like… If calling number begins with 516 True Call Redirect à 56789 False continue In CUCM… DN 56789 - CFwdAll to Voicemail - Unity CH or User w/ mailbox Add the partition of 56789 to the CSS of the UCCX CTI ports performing the redirect. - Dan From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Tuesday, March 10, 2015 12:23 PM To: Daniel Pagan; norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Daniel, There you go trying to make things slick and cool, lol! I like that approach! So, it seems his need is that if only 1 particular DNIS dialed one particular DN (a uccx trigger from my understanding) in ccm, that it then route the call to a specific VM account; but that it would also allow that DNIS to dial any other DN in the cluster without going to that specific VM account.. Being that I love to learn new things constantly; in your approach, is there a way to accomplish that requirement? Thanks, Ryan From: dpa...@fidelus.com To: norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 15:09:32 + Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 For routing based on ANI, Ryan Huff’s suggestion will certainly help from the perspective of CUCM. You can use this alongside CUC routing rules – route calls from the 519 area code to a specific mailbox. If you know these callers from area code 519 are dialing the same DNIS, then routing by ANI in CUCM won’t be needed – simply route the DNIS to voicemail, add a routing rule matching the 519 calls, and route to a mailbox. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of norm.nichol...@kitchener.ca Sent: Tuesday, March 10, 2015 8:43 AM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 I have been asked to send an outside caller to a voicemail box so when the DNIS or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is this possible ? Thanks Norm Nicholson Telecom Analyst City of Kitchener (519) 741-2200
Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
I'm not sure I correctly understand your question. Are you asking if instead of routing the calling number directly to a VM account, route to a UCCX queue trigger? Thanks, Ryan From: norm.nichol...@kitchener.ca To: ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Date: Tue, 10 Mar 2015 14:06:21 + Great info and one more question…. Can we do this on specific CCX pilot # s ( they are DID’s ) verses every call this number makes to our call manager?. Thanks From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Tuesday, March 10, 2015 9:26 AM To: Norm Nicholson; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 If you have a modern CCM version (8.x or above), which I'm guessing you are indicating such by the subject line of the email, you can create a scenario where CCM can perform routing decisions based upon the calling party number. Please review this post: http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ and follow the steps. The variation that you are going to need to do though is once CCM makes a match on the calling party number you'll need to translate the called party number to a CTI Route Point or something that forwards into the Unity VM account that you want. Thanks, Ryan From: ryanh...@outlook.com To: norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 08:57:27 -0400 Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Could you tell me what version of CCM you are dealing with? If you go under Call Routing - Translation Pattern and click Add New, do you see Route Next Hop By Calling Party Number? Thanks, Ryan From: norm.nichol...@kitchener.ca To: cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 12:43:04 + Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 I have been asked to send an outside caller to a voicemail box so when the DNIS or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is this possible ? Thanks Norm Nicholson Telecom Analyst City of Kitchener (519) 741-2200 x 7000 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Yes you can do that, but it gets a bit more tricky, and since your dealing with a CCX trigger, you could probably script something in uccx just as easy. If you want to do this in CCM though, put on a hard hat and lets get to it ... Assuming you've already read and deployed the previous example ... In the initial Partition that contains the ! pattern that routes everything by calling party, you would create another translation in that same partition and the pattern would be the called party number. So the partition would contain a translation for ! and a translation for the CCX trigger. Both translations would route to next hop by calling party. The translation that contains your CCX trigger would have to use a CSS that called a different partition than the ! translation. In the new partition there would be a ! translation pattern that would call out with a CSS that can access the DN's partition and then a translation pattern that matches the called party and then either block the route or translate the calling party to your CTI Route Point that forwards to your VM account in Unity. In the end, you end up with a scenario where the CCX trigger is called and routed to the next hop by calling party and once at the next hop, if the calling party matches the defined pattern, is blocked or routed. However, if that calling party called any other DN besides the CCX trigger, it would route to the next hop in the original filter partition and be allowed to call (provided I wasn't blocking that calling party in that partition). The reason all this works is because of CCM's route matching matrix; wherein the most specific route match always wins routing decisions. A specified number pattern will always be more specific than ! which essentially matches anything. Let me know if you need addition help offline, this is sort of complicated in writing but when you see it in production you're like really, that's it? Thanks, Ryan Huff From: norm.nichol...@kitchener.ca To: ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Date: Tue, 10 Mar 2015 15:00:48 + Sorry the incoming phone number we want to route to a voicemail box, but only when it calls in on a specific CCX pilot # . They should be allowed to call all other numbers and DID’s within our Call Manager. Thanks From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Tuesday, March 10, 2015 10:45 AM To: Norm Nicholson; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 I'm not sure I correctly understand your question. Are you asking if instead of routing the calling number directly to a VM account, route to a UCCX queue trigger? Thanks, Ryan From: norm.nichol...@kitchener.ca To: ryanh...@outlook.com; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Date: Tue, 10 Mar 2015 14:06:21 + Great info and one more question…. Can we do this on specific CCX pilot # s ( they are DID’s ) verses every call this number makes to our call manager?. Thanks From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Tuesday, March 10, 2015 9:26 AM To: Norm Nicholson; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 If you have a modern CCM version (8.x or above), which I'm guessing you are indicating such by the subject line of the email, you can create a scenario where CCM can perform routing decisions based upon the calling party number. Please review this post: http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ and follow the steps. The variation that you are going to need to do though is once CCM makes a match on the calling party number you'll need to translate the called party number to a CTI Route Point or something that forwards into the Unity VM account that you want. Thanks, Ryan From: ryanh...@outlook.com To: norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 08:57:27 -0400 Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 Could you tell me what version of CCM you are dealing with? If you go under Call Routing - Translation Pattern and click Add New, do you see Route Next Hop By Calling Party Number? Thanks, Ryan From: norm.nichol...@kitchener.ca To: cisco-voip@puck.nether.net Date: Tue, 10 Mar 2015 12:43:04 + Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0 I have been asked to send an outside caller to a voicemail box so when the DNIS or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is this possible ? Thanks Norm Nicholson Telecom Analyst City of Kitchener (519) 741-2200 x 7000
Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...
In your email below it seemed you were trying an update from 10.0.1 to 10.5.1 If you are coming FROM 9.1.2, you may be hitting this bug https://tools.cisco.com/bugsearch/bug/CSCup45923 On page 3 of the 9.1.2 SU2 release notes (http://www.cisco.com/web/software/282074295/113937/cucm-readme-912su2-Rev2.pdf) it warns of this bug. The reported work around is to go directly to 10.5.1. f you are suggesting that you are still receiving the unknown error when trying to directly upgrade 9.1.2 TO 10.5.1, then I'd go ahead and raise a TAC case. If this isn't a production cluster or I didn't mind taking it down, I'd reboot the cluster then try a 9.1.2 - 10.5.1 upgrade and if I still got the error, I'd raise a TAC case. Thanks, Ryan Date: Tue, 10 Mar 2015 09:41:10 -0500 Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails... From: jonv...@gmail.com To: ryanh...@outlook.com CC: agrec...@gmail.com; cisco-voip@puck.nether.net I applied the RSA cop file... the version from is 9.1.2.11900-12 I have tried going to: 10.0.1.11900-210.5.1.1-7 Same errors every time. All of them are the non-bootable ISO. Jonathan On Tue, Mar 10, 2015 at 6:22 AM, Ryan Huff ryanh...@outlook.com wrote: Jonathan, Have you applied any other cop or engineering special files besides the version 3 keys? Is your cluster based on the unrestricted export version or the restricted export version? Does the update your applying match (restricted versus unrestricted) Can you list out the specific version of 10.0.1 you are using and the specific version version of 10.5.1 that you want to go to? If you do a show version active on all the cluster nodes; what do you see? Is it the same on all nodes? I assume you tried to download a new image file from CCO and use that? Thanks, Ryan Original Message From: Jonathan Charles jonv...@gmail.com Sent: Tuesday, March 10, 2015 01:30 AM To: Andrew Grech agrec...@gmail.com Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails... CC: Cisco VoIP Group cisco-voip@puck.nether.net Well, that didn't work... SFTP'd the ISO to the box, same error... Same error message... Jonathan On Mon, Mar 9, 2015 at 11:45 PM, Jonathan Charles jonv...@gmail.com wrote: It is re-running after SFTPing from a Windows box (FreeFTPd)... Jonathan On Mon, Mar 9, 2015 at 11:31 PM, Andrew Grech agrec...@gmail.com wrote: Hi john, I've had a ISO from Linux to prime with bad permissions and had this error On 10/03/2015 1:13 PM, Jonathan Charles jonv...@gmail.com wrote: OK, let me try via SFTP (Windows box)... see what happens... Jonathan On Mon, Mar 9, 2015 at 9:26 PM, Charles Goldsmith wo...@justfamily.org wrote: Given that it is complaining about accessing the upgrade file, are you using dvd image/dvd on the host or sftp? I'd try a different method to see. Not a whole lot of information, but I have seen SFTP cause similar issues, I think it was a permissions problem on a linux setup using openssh. On Mon, Mar 9, 2015 at 8:16 PM, Jonathan Charles jonv...@gmail.com wrote: Getting error: Error encountered: An unknown error occurred while accessing the upgrade file. Warning: A system reboot is required when the upgrade process completes or is canceled. This will ensure services affected by the upgrade process are functioning properly. Tried upgrade from to 10.0.1 and 10.5.1, same errors... I have applied the ciscocm.version3-keys.cop.sgn patch, no change in behavior... Any ideas? Jonathan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] vMotion w/o Shared Storage
Shutdown, yes. Running, no. Thanks, Ryan Original Message From: Daniel Pagan dpa...@fidelus.com Sent: Tuesday, March 10, 2015 05:06 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] vMotion w/o Shared Storage Quick question... vMotion of a shut down UCM between two hosts without shared storage using vCenter. Is this supported? The virtualization document for UC platforms says vMotion is supported on shared storage, so I figured to ask. Is this migration method supported? - Dan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Sip design question
I have a pair of cubes on 4000 series ISRs. I want to do cube-ha on the ccm facing side and the itsp facing side. 1.) Am I better off just doing HSRP on both sides (which is 70% of cube-ha anyway) or is it practical to do the connected call failover portion? 2.) If I include the connected call failover, which side would I do that one, 1 or both (ccm facing side or itsp facing side)? Thanks, Ryan___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Temp Fail since upgrade
Lisa,
SIP or TDM/PRI?
Have you gandered into RTMT and taken a look at any active / recent alerts?
Thanks,
Ryan
Original Message
From: Lisa Notarianni lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade
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A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to
10.5.1.
We have experienced 2 intermittent outbound calling issues:
1. “Temp Fail” shows on phone display and only option on phone button is
“End Call”
2. Outbound callers dial a number and hear nothing but notice the phone
display says “Connected”. Caller cannot hear anything. Person they called
cannot hear anything.
***These issues are only intermittent.***
We had no issues before the upgrade.
Is anyone experiencing the similar issues?
Thank you in advance.

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Temp Fail since upgrade
I'm not aware of any known caveats of that nature regarding that upgrade path. Any recent network changes between ccm and the gateways since the upgrade? I have seen situations with MGCP where the ccm side of the gateway config had to be rebuilt after a CCM upgrade. Are you running an ED code version on the gateway? Check for caveats in your gateway code version with ccm 10.5.1 Have you restarted the MGCP process on the gateways (inside a maintenance window)? In a more drastic approach, you could try and reintegrate the gateways as h.323 and see if the issue persists. Thanks, Ryan Original Message From: Lisa Notarianni lisa.notaria...@scranton.edu Sent: Monday, March 30, 2015 05:01 PM To: Ryan Huff ryanh...@outlook.com,cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Temp Fail since upgrade 2 PRIs one each Communication Media Module– MGCP gateways. 2 Call Managers set up as redundant. Cisco TAC looked at RTMT with me today. They want us to set up packet captures and try to duplicate the problem tomorrow. I am just wondering if this is an upgrade issue that anyone else may be experiencing since we had no problems before the upgrade. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Monday, March 30, 2015 4:52 PM To: Lisa Notarianni; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Temp Fail since upgrade Lisa, SIP or TDM/PRI? Have you gandered into RTMT and taken a look at any active / recent alerts? Thanks, Ryan Original Message From: Lisa Notarianni lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu Sent: Monday, March 30, 2015 04:48 PM To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: [cisco-voip] Temp Fail since upgrade A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 10.5.1. We have experienced 2 intermittent outbound calling issues: 1. “Temp Fail” shows on phone display and only option on phone button is “End Call” 2. Outbound callers dial a number and hear nothing but notice the phone display says “Connected”. Caller cannot hear anything. Person they called cannot hear anything. ***These issues are only intermittent.*** We had no issues before the upgrade. Is anyone experiencing the similar issues? Thank you in advance. [LNsignatureFile] ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Sip design question
Charles, I guess that is a better place to start; I may be going down this road in a near future. I have been reading http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-high-availability.html#concept_5013D60352C446769D62736C8CDE87E8 which seems to suggest that L2 box to box is possible on the 4451. Are you saying it is not? Thanks, Ryan From: wo...@justfamily.org Date: Tue, 31 Mar 2015 23:59:34 -0600 Subject: Re: [cisco-voip] Sip design question To: ryanh...@outlook.com CC: cisco-voip@puck.nether.net Please correct me if I'm wrong, but I thought cube-ha was missing from the code on these? same as the ASR's since they are all running ios-xe. I have not tested it myself, just doing a lot of reading in preparation of deploying these. On Mon, Mar 30, 2015 at 6:29 AM, Ryan Huff ryanh...@outlook.com wrote: I have a pair of cubes on 4000 series ISRs. I want to do cube-ha on the ccm facing side and the itsp facing side. 1.) Am I better off just doing HSRP on both sides (which is 70% of cube-ha anyway) or is it practical to do the connected call failover portion? 2.) If I include the connected call failover, which side would I do that one, 1 or both (ccm facing side or itsp facing side)? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2
I've used the ATA190 and it's family of 18X pretty often. In the case of the SIP devices like the 190 or 187 I have always been able to use the default SIP Profile settings; of course I'm generally dealing with vanilla networks and CCM's built very closely to the SRND. Here is where I would start (if you haven't already): What do you have the device security mode set to? If you have it set to Authenticated or Encrypted but it has not received a CTL file, the phone will attempt registration up to four times to make a secure connection. Are the MAC addresses for Phone 1 and Phone 2 correct (Phone 2 will usually loose the first two characters and append a 01 at the end of the device name). If you support auto registration, remove the devices and let them auto register and see if they stabilize. Any layer 1/2 issues (change out the patch cable, check the connection into the switchport ... etc)? Do you see any interface drops on the switchport? Is the switchport full duplex 10/100 or 10/1000? Have you shut/no shut the port just to see if clearing the port helps? Go through the IVR menu on the ATA itself and verify that it reports all the same settings that you CCM device configs show. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/190/1_0/english/administration/guide/sip/ATA190/a190_agBcd.html Thanks, Ryan From: bertacco.alessan...@alice.it To: cisco-voip@puck.nether.net Date: Mon, 23 Mar 2015 11:28:44 +0100 Subject: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2 Hi Guys, ATA190 FW version 1.1.2.(005) with CUCM 10.5.2.1-5, de-register randomly, and need to reboot the device to re-register again on the CUCM. Anyone as the same issue? Do you use a special SIP Device profile changing some timings, or you are using standard SIP Device Profile? Thank you all Regards Alessandro Bertacco ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2
I should also add if you have a spare ATA190, swap it out and see if the issue stabilizes or continues on with the different device. If the issue persists with a different device, likely not the ATA. If you don't have a spare 190, then try your 190 on a different cluster if you have one (I don't think the 190 is officially supported on any version of CME yet, but you could try that if you don't have another cluster). Thanks, Ryan From: ryanh...@outlook.com To: bertacco.alessan...@alice.it; cisco-voip@puck.nether.net Date: Mon, 23 Mar 2015 09:19:36 -0400 Subject: Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2 I've used the ATA190 and it's family of 18X pretty often. In the case of the SIP devices like the 190 or 187 I have always been able to use the default SIP Profile settings; of course I'm generally dealing with vanilla networks and CCM's built very closely to the SRND. Here is where I would start (if you haven't already): What do you have the device security mode set to? If you have it set to Authenticated or Encrypted but it has not received a CTL file, the phone will attempt registration up to four times to make a secure connection. Are the MAC addresses for Phone 1 and Phone 2 correct (Phone 2 will usually loose the first two characters and append a 01 at the end of the device name). If you support auto registration, remove the devices and let them auto register and see if they stabilize. Any layer 1/2 issues (change out the patch cable, check the connection into the switchport ... etc)? Do you see any interface drops on the switchport? Is the switchport full duplex 10/100 or 10/1000? Have you shut/no shut the port just to see if clearing the port helps? Go through the IVR menu on the ATA itself and verify that it reports all the same settings that you CCM device configs show. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/190/1_0/english/administration/guide/sip/ATA190/a190_agBcd.html Thanks, Ryan From: bertacco.alessan...@alice.it To: cisco-voip@puck.nether.net Date: Mon, 23 Mar 2015 11:28:44 +0100 Subject: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2 Hi Guys, ATA190 FW version 1.1.2.(005) with CUCM 10.5.2.1-5, de-register randomly, and need to reboot the device to re-register again on the CUCM. Anyone as the same issue? Do you use a special SIP Device profile changing some timings, or you are using standard SIP Device Profile? Thank you all Regards Alessandro Bertacco ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Sip design question
CUBE-HA on a 4k doesn't seem very battle tested yet. Clearly it shouldn't go the way of CUBE-SP on an ASR1k which got dumped. Some of those are significant caveats though (SDP passthru being a possible deal killer for me); almost makes just doing plain old HSRP and setting the client expectation for failover seem just as reasonable. Thanks, Ryan From: wo...@justfamily.org Date: Wed, 1 Apr 2015 08:17:45 -0600 Subject: Re: [cisco-voip] Sip design question To: ryanh...@outlook.com CC: cisco-voip@puck.nether.net Per this: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_mgmt/configuration/xe-3s/cube-mgmt-xe-3s-book/voi-stateful-switchover.html it says it is on 3.2 or later, but it does have a list of caveats, perhaps that is what I was thinking about. Sorry for the false alarm. On Wed, Apr 1, 2015 at 5:33 AM, Ryan Huff ryanh...@outlook.com wrote: Charles, I guess that is a better place to start; I may be going down this road in a near future. I have been reading http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-high-availability.html#concept_5013D60352C446769D62736C8CDE87E8 which seems to suggest that L2 box to box is possible on the 4451. Are you saying it is not? Thanks, Ryan From: wo...@justfamily.org Date: Tue, 31 Mar 2015 23:59:34 -0600 Subject: Re: [cisco-voip] Sip design question To: ryanh...@outlook.com CC: cisco-voip@puck.nether.net Please correct me if I'm wrong, but I thought cube-ha was missing from the code on these? same as the ASR's since they are all running ios-xe. I have not tested it myself, just doing a lot of reading in preparation of deploying these. On Mon, Mar 30, 2015 at 6:29 AM, Ryan Huff ryanh...@outlook.com wrote: I have a pair of cubes on 4000 series ISRs. I want to do cube-ha on the ccm facing side and the itsp facing side. 1.) Am I better off just doing HSRP on both sides (which is 70% of cube-ha anyway) or is it practical to do the connected call failover portion? 2.) If I include the connected call failover, which side would I do that one, 1 or both (ccm facing side or itsp facing side)? Thanks, Ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] FXS voltages / POTS compatibility
Lelio, You could remove the mgcp service from the fxs port, then create your dial peers (assuming your alarm only wants tone and doesn't need inward). Thanks, Ryan From: le...@uoguelph.ca Date: Mon, 2 Mar 2015 08:21:51 -0500 To: jandrewar...@ccgs.wa.edu.au CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] FXS voltages / POTS compatibility James, Does this mean you have an h323 gateway? Right now, I have MGCP, which I'm guessing, precludes me from doing this. Sent from my iPhone On Mar 2, 2015, at 7:51 AM, James Andrewartha jandrewar...@ccgs.wa.edu.au wrote: With our security systems I have to remove the call manager from the call path for the system to complete due to the nonstandard tones they send. From my notes on how to configure this: On the VG224s: voice class h323 1 h225 timeout tcp establish 3voice-port 2/13 no timeoutsdial-peer voice 23 pots service stcappdial-peer voice 99 voip description h323 direct to voip1 for alarm number destination-pattern 13451015 session target ipv4:10.101.0.5 voice-class h323 1 codec g711ulaw no vaddial-peer voice 98 voip description h323 direct to voip2 for alarm number preference 1 destination-pattern 13451015 session target ipv4:10.101.0.6 voice-class h323 1 codec g711ulaw no vad On the 2921s: voice service voipip address trusted list ipv4 10.100.0.10 255.255.255.255 The AVG is 10.100.0.10, the 2921s are 10.101.0.5 and .6, and dial-peer voice 23 is for voice-port 2/13. These are GE security panels I think (which their MAC OUI confirms). -- James AndrewarthaNetwork Projects EngineerChrist Church Grammar SchoolClaremont, Western AustraliaPh. (08) 9442 1757Mob. 0424 160 877 From: Justin Steinberg jsteinb...@gmail.com Date: Monday, 2 March 2015 2:14 am To: chris tknch...@gmail.com Cc: Cisco VOIP cisco-voip@puck.nether.net Subject: Re: [cisco-voip] FXS voltages / POTS compatibility Are you using local h323 or sip 'pots' dialpeers to route directly between your FXS and T1 port? Or is call manager in between the call due to MGCP or VOIP dialpeers involved in the dialplan ?I doubt your issue is line voltage, since you can see the call being placed. My guess is the DSP is processing the call and causing issues. I've seem alarm boxes use nonstandard DTMF transmission that isn't properly recognized by the DSP.The 2800 supports DSP bypass by default when you route directly between ports using POTS dialpeers. You do need to have properly configured network clock configuration.Can you send a copy of your config along with the output of 'show controller t1' and 'show network-clock'JustinOn Feb 28, 2015 10:33 PM, chris tknch...@gmail.com wrote: Hey Ryan, We have a channelized T1 with channels split between voice/data so the voice path is TDM. We have a VIC-4FXS/DID and for each of the two ports we have a single copper pair with rj11 on both ends, one side going to the FXS port and the other is going into alarm panel. The total distance from the 2800 to the alarm panel is around 20-30 feet and its a direct run, no 66 blocks or anything in between.Don't know model of the panel (this is another location) From what I've read I think the problem is the default idle-voltage the VIC-4FXS/DID is only -24V but based on the link I sent in the first email I thought this could be reconfigured through the idle-voltage option but this doesnt seem to be available when I try to enter it under the voice-port. When I talked to the alarm company and told them I see the calls going through the guy told me the alarm doesn't check the line state based on the dialtone and he said that it uses the voltage to see when the line is idle, ringing, etc and I think this is where the problem lies. Someone recommend this adapter offlist which looks interesting but the price is a little nuts as it costs more than all the equipment we have installed at this site combined. http://www.homedepot.com/p/Viking-1-Line-Long-Loop-Adapter-VK-LLA-1/20435 Chris On Sat, Feb 28, 2015 at 8:18 PM, Ryan Huff ryanh...@outlook.com wrote: Chris,Can you diagram the connections for me? Are the copper pairs swinging off a 66 block before terminating to the alarm panel or is there a direct copper path between the fxs port and the alarm? Are you using an RJ-11 or RJ-14 configuration?Could you estimate the copper distance between the termination points?Is the pstn path for the VG SIP or TOM?Also, I would be curious to know if the alarm panel is a Simplex Grinnell?Thanks,Ryan Original Message From: chris tknch...@gmail.com Sent: Saturday, February 28, 2015 07:52 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] FXS voltages / POTS compatibility HelloWe have a location with a 2800 acting as a voice gateway where we have 2 FXS voice ports going to an alarm system. We are using the vic-4fxs/did line card.We have the alarm company saying they are seeing the panel reporting Comm trouble so we checked the call records
Re: [cisco-voip] FXS voltages / POTS compatibility
Lelio,
You could do that, but you would have to split out the channels on your T1.
Thanks,
Ryan
Original Message
From: Lelio Fulgenzi le...@uoguelph.ca
Sent: Monday, March 2, 2015 10:26 AM
To: Ryan Huff ryanh...@outlook.com
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility
CC: Cisco VOIP cisco-voip@puck.nether.net,James Andrewartha
jandrewar...@ccgs.wa.edu.au
p { margin: 0; }
OK, so have a few FXS ports reserved for alarm outbound calling. That could
work, but I was hoping to capitalize on the existing PRI connection.
---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph
519‐824‐4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1
From: Ryan Huff ryanh...@outlook.com
To: Lelio Fulgenzi le...@uoguelph.ca, James Andrewartha
jandrewar...@ccgs.wa.edu.au
Cc: Cisco VOIP cisco-voip@puck.nether.net
Sent: Monday, March 2, 2015 8:48:20 AM
Subject: RE: [cisco-voip] FXS voltages / POTS compatibility
!-- .hmmessage P { margin:0px; padding:0px } body.hmmessage { font-size:
12pt; font-family:Calibri } --
Lelio,
You could remove the mgcp service from the fxs port, then create your dial
peers (assuming your alarm only wants tone and doesn't need inward).
Thanks,
Ryan
From: le...@uoguelph.ca
Date: Mon, 2 Mar 2015 08:21:51 -0500
To: jandrewar...@ccgs.wa.edu.au
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility
James,
Does this mean you have an h323 gateway? Right now, I have MGCP, which I'm
guessing, precludes me from doing this.
Sent from my iPhone
On Mar 2, 2015, at 7:51 AM, James Andrewartha jandrewar...@ccgs.wa.edu.au
wrote:
With our security systems I have to remove the call manager from the call path
for the system to complete due to the nonstandard tones they send. From my
notes on how to configure this:
On the VG224s:
voice class h323 1
h225 timeout tcp establish 3
voice-port 2/13
no timeouts
dial-peer voice 23 pots
service stcapp
dial-peer voice 99 voip
description h323 direct to voip1 for alarm number
destination-pattern 13451015
session target ipv4:10.101.0.5
voice-class h323 1
codec g711ulaw
no vad
dial-peer voice 98 voip
description h323 direct to voip2 for alarm number
preference 1
destination-pattern 13451015
session target ipv4:10.101.0.6
voice-class h323 1
codec g711ulaw
no vad
On the 2921s:
voice service voip
ip address trusted list
ipv4 10.100.0.10 255.255.255.255
The AVG is 10.100.0.10, the 2921s are 10.101.0.5 and .6, and dial-peer voice
23 is for voice-port 2/13. These are GE security panels I think (which their
MAC OUI confirms).
--
James Andrewartha
Network Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
From: Justin Steinberg jsteinb...@gmail.com
Date: Monday, 2 March 2015 2:14 am
To: chris tknch...@gmail.com
Cc: Cisco VOIP cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility
Are you using local h323 or sip 'pots' dialpeers to route directly between
your FXS and T1 port? Or is call manager in between the call due to MGCP or
VOIP dialpeers involved in the dialplan ?
I doubt your issue is line voltage, since you can see the call being placed.
My guess is the DSP is processing the call and causing issues. I've seem
alarm boxes use nonstandard DTMF transmission that isn't properly recognized
by the DSP.
The 2800 supports DSP bypass by default when you route directly between ports
using POTS dialpeers. You do need to have properly configured network clock
configuration.
Can you send a copy of your config along with the output of 'show controller
t1' and 'show network-clock'
Justin
On Feb 28, 2015 10:33 PM, chris tknch...@gmail.com wrote:
Hey Ryan,
We have a channelized T1 with channels split between voice/data so the voice
path is TDM. We have a VIC-4FXS/DID and for each of the two ports we have a
single copper pair with rj11 on both ends, one side going to the FXS port and
the other is going into alarm panel. The total distance from the 2800 to the
alarm panel is around 20-30 feet and its a direct run, no 66 blocks or
anything in between.
Don't know model of the panel (this is another location)
From what I've read I think the problem is the default idle-voltage the
VIC-4FXS/DID is only -24V but based on the link I sent in the first email I
thought this could be reconfigured through the idle-voltage option but this
doesnt seem to be available when I try to enter it under the voice-port. When
I talked to the alarm company and told them I see the calls going through the
guy told me the alarm doesn't check the line state based on the dialtone and
he said that it uses the voltage to see when the line is idle, ringing, etc
and I think this is where
Re: [cisco-voip] errors with IOS transcoder
So in order to configure the transcoder, you must first initialize the DSP Farm. If you plan to use the dspfarm with your T1 card, then yes, you would need the card installed first. If you are not going to use the dspfarm with your T1 card, then you need to initialize the dspfarm on the voice-card that the DSP shows up on (doing a show inventory should reveal that to you). Thanks, Ryan Date: Tue, 3 Mar 2015 12:07:25 -0500 From: bhowser5...@gmail.com To: cisco-voip@puck.nether.net Subject: [cisco-voip] errors with IOS transcoder I am getting ready to add a VWIC3 T1 (and PRI) to my 2900 series ISR router running 15.3(2) and I'm trying to configure an IOS transcoder, I have PVDM3-64 (2 32 modules). When I get into the transcode sub configuration it gives me an Unrecognised Command error when trying to preference the codecs. What I am doing wrong? router(config)#dspfarm profile 6 transcode router(config-dspfarm-profile)#codec ? % Unrecognized command router(config-dspfarm-profile)#codec I've been reading that I may need to actually have the T1 card in the router first but I can't find anything to confirm that. I've been reading here: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cminterop/configuration/15-mt/dia-15-mt-book/vc-enh-confr-vgr.html#GUID-2E7DACC4-090A-4E89-BC1B-006E79BF0BC0 which seems to suggest that PVDM3 might not be able to support transcoding? Can anyone shed a little light my way? thank you ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] FXS voltages / POTS compatibility
Chris, Can you diagram the connections for me? Are the copper pairs swinging off a 66 block before terminating to the alarm panel or is there a direct copper path between the fxs port and the alarm? Are you using an RJ-11 or RJ-14 configuration? Could you estimate the copper distance between the termination points? Is the pstn path for the VG SIP or TOM? Also, I would be curious to know if the alarm panel is a Simplex Grinnell? Thanks, Ryan___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Auto Attendant Web Access Imges Issue
Are talking about attendant console or a unity connection system call handler? I assume you're talking about attendant console; did you double check your IIS setup from the deployment docs? Thanks, Ryan Original Message From: AbdusSaboor Khan saboor.k...@gmail.com Sent: Sunday, February 22, 2015 02:28 PM To: Cisco VoIP List cisco-voip@puck.nether.net Subject: [cisco-voip] Auto Attendant Web Access Imges Issue Hi, Can anyone tell me what I am missing here , I have install the Auto Attendant, as it says if you dont have SQL server willl install the express version, but install the 32bit version of standard SQL, so far everything goes well while I am accessing through webpage i am getting everything on the webpage except no images is coming which shows i am missing something in setup, so web page without images. Thanks if someone reply me quick will be great. Abdul ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco unity connection call handler
Not sure which version of CUC you're dealing with but this is a link to the System Administration Guide for Cisco Unity Connection Release 10.x http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/guide/10xcucsagx.html If you can explain the specific issue you're having, this list has a wealth of resources that could assist with CUC and system call handlers. Thanks, Ryan From: claitoncam...@gmail.com Date: Mon, 23 Feb 2015 01:08:34 + To: cisco-voip@puck.nether.net Subject: [cisco-voip] Cisco unity connection call handler Anyone with knowledge in unity connection call handler? If you can show me a document. I am implementing but I have had some problems. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Add new vCPU of CUCM server
Hmmm .. Did you add the correct vCPU amount back? Also, has the cluster/node been rebooted since adding the vCPU back? Thanks, Ryan Original Message From: Claiton Campos claitoncam...@gmail.com Sent: Monday, February 23, 2015 06:41 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Add new vCPU of CUCM server Hello , someone has come to change the amount of vCPU a CUCM server in production? Recently 'm having problems with high throughput publisher server and as directed by the TAC is suggested to add a new vCPU to the server. My question is whether the server recognizes this new vCPU . ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CCNA/CCNP Collaboration certs
Generally when Cisco does this, the previous credential is retired and you cannot get new certs in the previous credential after the retire date, only the new credential can be attained at that point. For those with a valid previous credential, they remain valid so long as the practitioner keeps up with a valid recertification process (via other tests). If the recertification period lapses, then the previous credential expires and is not recoverable and the practitioner would have to undergo the new credential certification process (in completeness). Often, you'll see practitioners keep the previous credential through recertification and attain the new credential, to have dual of something. You can also take a (usually smaller) migration path and just upgrade the credential you have into the new credential. Thanks, Ryan Original Message From: zoltan.kele...@emerson.com Sent: Tuesday, February 24, 2015 03:35 AM To: ryanh...@outlook.com,ben.st...@gmail.com,wo...@justfamily.org Subject: RE: [cisco-voip] CCNA/CCNP Collaboration certs CC: cisco-voip@puck.nether.net They are just retiring, not actually *invalidating* CCNP Voice, right? I mean all of our CCNP Voice certificates won't go up in a puff of smoke once CCNP Collab comes out. Only that no new CCNP Voice certs would be awarded after a yet to be announced date (or has it been announced already? Anyway) Since I've just got mine, I don't think I'll be in any rush to update to Collaboration, especially not until at least some valuable training materials are out . From my experience, I wouldn't rely on classroom training alone. So far none of the official cisco training classes I've attended covered *all* of the exam material. And there's barely a synopsis for CIPTV2 out so far. Cheers, Zoltan Kelemen Global Communications and Information Security Implementation Engineering Emerson From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: 17 February 2015 19:58 To: Ben Story; Charles Goldsmith Cc: voip puck Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs What gets me is that a CCNP Voice needs one test to upgrade to CCNP Collaboration (http://www.cisco.com/web/learning/tools/ccnp_collab/ccnp_collab_tool.html). Really? Only one test's worth of info is what changes between Voice and Collaboration? If you are going to retire and invalidate a credential and move to another tract altogether, then it should be a top-down revamp, not just one test of a few new questions about video. This feels like it is nothing more than re branding/marketing; and I shouldn't have to pay for that. My grumpy two cents, Ryan Date: Tue, 17 Feb 2015 10:10:24 -0600 From: ben.st...@gmail.commailto:ben.st...@gmail.com To: wo...@justfamily.orgmailto:wo...@justfamily.org CC: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs Is it just me or does it seem weird for a CCNA level to now require two tests. Money grab? -- Ben Story CCNP, CCNA, CCNA Wireless, CCDA ben.st...@gmail.commailto:ben.st...@gmail.com mailto:ben.st...@gmail.com @ntwrk80 http://showbrain.blogspot.com http://rand0mw0rds.blogspot.comhttp://rand0mw0rds.blogspot.com/ From sour-faced saints and silly devotions, good Lord, preserve us!. -- St. Teresa of Avila On Tue, Feb 17, 2015 at 9:58 AM, Charles Goldsmith wo...@justfamily.orgmailto:wo...@justfamily.org wrote: https://learningnetwork.cisco.com/community/ccna-ccnp-collaboration ___ cisco-voip mailing list cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] strange t1 issue
Doing a PRI migration from a 2811 vwic2 T1 to a 2911 vwic3 T1 I can get the D Channel (Serial 0/1/0:23) to come up but I can't get the B channels to come up. The Only thing in the logs are: %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 10 to priority 1 : %LINK-3-UPDOWN: Interface Serial0/1/0:23, changed state to up : %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 1 to priority 10 : %LINK-3-UPDOWN: Interface Serial0/1/0:23, changed state to down : %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 10 to priority 1 : %LINK-3-UPDOWN: Interface Serial0/1/0:23, changed state to up Every so often it repeats. Any ideas thanks, ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] strange t1 issue
Phillip, The d channel comes up but the b's will not. Thanks, Ryan From: ryanh...@outlook.com To: philip.wale...@polycom.com Date: Wed, 25 Feb 2015 19:39:13 -0500 CC: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] strange t1 issue Phillip, T1 0/1/0 is up. Applique type is Channelized T1 Cablelength is long 0db Description: ATT SE Circuit 10.IPZX.518281.001.SB 800-247-2020 No alarms detected. alarm-trigger is not set Soaking time: 3, Clearance time: 10 AIS State:Clear LOS State:Clear LOF State:Clear Version info FPGA Rev: 08121917, FPGA Type: PRK4 Framing is ESF, Line Code is B8ZS, Clock Source is Line. CRC Threshold is 320. Reported from firmware is 320. Data in current interval (235 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs Data in Interval 1: 0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 90 Unavail Secs Total Data (last 1 15 minute intervals): 0 Line Code Violations, 0 Path Code Violations, 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins, 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 90 Unavail Secs Thanks, Ryan From: philip.wale...@polycom.com To: ryanh...@outlook.com CC: cisco-voip@puck.nether.net Date: Wed, 25 Feb 2015 16:25:17 -0800 Subject: Re: [cisco-voip] strange t1 issue Do you have the clock sourced to a specific t1? It sounds like the clock isn't syncing correctly. What do the t1 counters show? Sent from my iPhone On Feb 25, 2015, at 5:27 PM, Ryan Huff ryanh...@outlook.com wrote: Doing a PRI migration from a 2811 vwic2 T1 to a 2911 vwic3 T1 I can get the D Channel (Serial 0/1/0:23) to come up but I can't get the B channels to come up. The Only thing in the logs are: %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 10 to priority 1 : %LINK-3-UPDOWN: Interface Serial0/1/0:23, changed state to up : %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 1 to priority 10 : %LINK-3-UPDOWN: Interface Serial0/1/0:23, changed state to down : %MARS_NETCLK-3-CLK_TRANS: Network clock source transitioned from priority 10 to priority 1 : %LINK-3-UPDOWN: Interface Serial0/1/0:23, changed state to up Every so often it repeats. Any ideas thanks, ryan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] R: ATA190 Random De-Register from CUCM 10.5.2
Can you hop into CLI and show me the output of show version active.
Also, make sure that output is the same on all cluster nodes.
Thanks,
Ryan
Original Message
From: Alessandro Bertacco bertacco.alessan...@alice.it
Sent: Monday, March 23, 2015 04:10 PM
To: 'Rob Dawson' rdaw...@force3.com,'Ryan Huff'
ryanh...@outlook.com,cisco-voip@puck.nether.net
Subject: R: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2
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Hi Rob, in my case all the two lines are configured and used!
Regards
Alessandro
Da: Rob Dawson [mailto:rdaw...@force3.com]
Inviato: lunedì 23 marzo 2015 15:47
A: Ryan Huff; Alessandro Bertacco; cisco-voip@puck.nether.net
Oggetto: RE: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2
There used to be a defect where ATAs with only one line configured/registered
would deregister due to the second line constantly attempting to download a
configuration. I am not sure if it only impacted
Re: [cisco-voip] Cisco 8851 not failing over to backup circuit...
What load on the 51s and is that load version present on all nodes? Thanks, Ryan Original Message From: Jonathan Charles jonv...@gmail.com Sent: Thursday, April 23, 2015 08:44 PM To: Ryan Huff ryanh...@outlook.com Subject: Re: [cisco-voip] Cisco 8851 not failing over to backup circuit... CC: cisco-voip@puck.nether.net Settings are identical... I have no idea why only the 8831 is registering and the 8851s aren't... Jonathan On Thu, Apr 23, 2015 at 7:37 PM, Ryan Huff ryanh...@outlook.com wrote: Same ccm group/device pool etc? Thanks, Ryan Original Message From: Jonathan Charles jonv...@gmail.com Sent: Thursday, April 23, 2015 08:35 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Cisco 8851 not failing over to backup circuit... We have CUCM 8.6.2 with Cisco 8851, Cisco 8831 phones at a remote location; they are connected over MPLS and a Peplink Balance VPN as a backup. When we yank the MPLS, the 8831 registers with CUCM and works fine the 8851s do NOT. Any reason the 8851 would act differently? Jonathan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco 8851 not failing over to backup circuit...
Could also be a CTL issue if your doing secure ... Thanks, Ryan Original Message From: Ryan Huff ryanh...@outlook.com Sent: Thursday, April 23, 2015 08:48 PM To: Jonathan Charles jonv...@gmail.com Subject: Re: [cisco-voip] Cisco 8851 not failing over to backup circuit... CC: cisco-voip@puck.nether.net What load on the 51s and is that load version present on all nodes? Thanks, Ryan Original Message From: Jonathan Charles jonv...@gmail.com Sent: Thursday, April 23, 2015 08:44 PM To: Ryan Huff ryanh...@outlook.com Subject: Re: [cisco-voip] Cisco 8851 not failing over to backup circuit... CC: cisco-voip@puck.nether.net Settings are identical... I have no idea why only the 8831 is registering and the 8851s aren't... Jonathan On Thu, Apr 23, 2015 at 7:37 PM, Ryan Huff ryanh...@outlook.com wrote: Same ccm group/device pool etc? Thanks, Ryan Original Message From: Jonathan Charles jonv...@gmail.com Sent: Thursday, April 23, 2015 08:35 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Cisco 8851 not failing over to backup circuit... We have CUCM 8.6.2 with Cisco 8851, Cisco 8831 phones at a remote location; they are connected over MPLS and a Peplink Balance VPN as a backup. When we yank the MPLS, the 8831 registers with CUCM and works fine the 8851s do NOT. Any reason the 8851 would act differently? Jonathan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] QOS - Looking for another set of eyeballs
The below OUT map, applied in the output direction on a WAN(mpls) facing interface, should put RTP, Signaling and anything from access-list 51 at the top of the heap and give everything else best effort. Not that anything isn't working, I just want to make sure I'm not making something up ... etc. Seems basic but I don't get to play with QOS everyday :) router-3925#sh run | sec class-map|policy-map|access-list 51 ! ! class-map match-all VOICE match ip dscp ef match access-group 51 class-map match-any CALL-SIGNALING match ip dscp cs3 match ip dscp af31 ! ! policy-map WAN-OUT class VOICE bandwidth percent 30 class CALL-SIGNALING bandwidth percent 10 class class-default fair-queue ! ! access-list 51 permit 001.002.003.004 0.0.0.255 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Phone Restart on Subscriber add
So you can add the node without causing a phone reboot. Just keep it out of the
ccm group in the cluster. Then add it in during a maintrnance window.
Thanks,
Ryan
Original Message
From: Tommy Schlotterer tschlotte...@netechcorp.com
Sent: Monday, April 27, 2015 02:07 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Phone Restart on Subscriber add
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All,
I have a customer that is very picky on when phones will reboot when adding a
new subscriber to a CUCM cluster.
Does anyone know in what point the phones reboot? Right after the conntivity
vierfication? Right when the new node is done installing?
Thanks
Tommy
Tommy Schlotterer | Systems Engineer
CCNA, CCNA Voice
48325 Alpha Dr. Ste. 150
Wixom, MI 48393
p 248.468.0710
e tschlotte...@netechcorp.com
w netechcorp.com

   
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Re: [cisco-voip] RTMT??? or some other way to monitor a phone
Might be easier to start the other way around. What is actually wrong? RTMT does work with registered phones, and presumably, your wireless phones are registered? Are the wireless phones not registering to call manager? Thanks, Ryan Original Message From: Scott Voll svoll.v...@gmail.com Sent: Thursday, May 7, 2015 05:34 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] RTMT??? or some other way to monitor a phone We are troubleshooting a wireless deployment of 8861 phones. is there any way to monitor just a phone / phone model / device pool / etc so I can just see what I'm looking for? Looks like RTMT is just all registered phones. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] RTMT??? or some other way to monitor a phone
You can pull the sdl traces off the node the phone is registered with for a time range and it will show the rev/unreg events. In my experience with wireless phone deployments, sometimes users blur the lines between unregistered and not with a coverage area. Which are to very different things. Thanks, Ryan Original Message From: Scott Voll svoll.v...@gmail.com Sent: Thursday, May 7, 2015 05:46 PM To: Ryan Huff ryanh...@outlook.com Subject: Re: [cisco-voip] RTMT??? or some other way to monitor a phone CC: Brian Meade bmead...@vt.edu,cisco-voip@puck.nether.net my 8861 phones are working wirelessly just fine. I have a pilot of about 25 phones. The problem is I have A user who is complaining that the phone unregisters and registers from time to time. We want to see how often it's happening. Does that make sense? Scott On Thu, May 7, 2015 at 2:42 PM, Ryan Huff ryanh...@outlook.com wrote: So if ccm doesnt show phone registration AT ALL for your wireless phones, and auto registration is enabled/works or you have pre-built devices in ccm for the phones, I would start troubleshooting you're wireless gear. Thanks, Ryan Original Message From: Scott Voll svoll.v...@gmail.com Sent: Thursday, May 7, 2015 05:39 PM To: Brian Meade bmead...@vt.edu Subject: Re: [cisco-voip] RTMT??? or some other way to monitor a phone CC: cisco-voip@puck.nether.net Sorry, yes. I just want to see the registration of those pilot phones. Scott On Thu, May 7, 2015 at 2:38 PM, Brian Meade bmead...@vt.edu wrote: What exactly are you trying to monitor? Just registration status? On Thu, May 7, 2015 at 5:34 PM, Scott Voll svoll.v...@gmail.com wrote: We are troubleshooting a wireless deployment of 8861 phones. is there any way to monitor just a phone / phone model / device pool / etc so I can just see what I'm looking for? Looks like RTMT is just all registered phones. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] RTMT??? or some other way to monitor a phone
So if ccm doesnt show phone registration AT ALL for your wireless phones, and auto registration is enabled/works or you have pre-built devices in ccm for the phones, I would start troubleshooting you're wireless gear. Thanks, Ryan Original Message From: Scott Voll svoll.v...@gmail.com Sent: Thursday, May 7, 2015 05:39 PM To: Brian Meade bmead...@vt.edu Subject: Re: [cisco-voip] RTMT??? or some other way to monitor a phone CC: cisco-voip@puck.nether.net Sorry, yes. I just want to see the registration of those pilot phones. Scott On Thu, May 7, 2015 at 2:38 PM, Brian Meade bmead...@vt.edu wrote: What exactly are you trying to monitor? Just registration status? On Thu, May 7, 2015 at 5:34 PM, Scott Voll svoll.v...@gmail.com wrote: We are troubleshooting a wireless deployment of 8861 phones. is there any way to monitor just a phone / phone model / device pool / etc so I can just see what I'm looking for? Looks like RTMT is just all registered phones. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
